Call management system with call control from user workstation computers

ABSTRACT

A call management method and system. The system includes at least one user position, comprising a computer workstation and a telephone apparatus that is associated with the computer workstation. In addition, the system includes a call management computer comprising a memory; and a digital data network to connect the computer workstation with the call management computer. The memory is used to store a plurality of call processing rules that determine how a call, directed to a user, is to be processed. The plurality of call processing rules is defined by the computer workstation before the call is received. The call management computer intercepts the call, that is incoming, to a first user position that is included in the at least one user position. The call management computer determines that the call is for the first user position and interacts with the memory to determine how the call is processed based on the plurality of call processing rules. Finally, the call management computer processes the call according to instructions of at least one applicable call processing rule that is included in the plurality of call processing rules.

RELATED APPLICATIONS

This application is a divisional of U.S. application Ser. No.09/360,719, filed Jul. 27, 1999 now U.S. Pat. No. 7,136,475, which is adivisional of U.S. application Ser. No. 08/642,171, filed Mar. 11, 1996now U.S. Pat. No. 5,946,386, which both are herein incorporated byreference.

BACKGROUND OF THE INVENTION

This invention pertains to telephone switching systems in general.

Business communication has taken two separate paths. One involvestelephone conversations and the other involving computer communication.

Until now, business telephone communications have been based upon theapproach that each individual controls his own call traffic throughmultiple buttons on proprietary telephone instruments and/or simplecommands entered through “hookflash” or the telephone keypad. Further,the architecture and philosophy applied to business PBXs or othertelephone switches is limited to the “switching” of calls, such asincoming calls, to internal stations or internal stations to internalstations. This approach strictly avoids operation based upon “callcontent” such as the type of call, from whom it originates, etc. Thelimited capabilities of the multi-button telephone instruments and thelack of awareness of call content severely restrict the capabilities andfeatures available and thus reduce the overall effectiveness of thebusiness telephone systems of the past.

The focus of computer technology has become the desktop workstationcomputer attached to one or more business enterprise-wide, high-speeddigital networks which interconnect the workstation computers ofbusiness enterprise's employees with a variety of information servers,communications and computing devices. The business enterprise's digitalnetwork may be a combination of Local Area Networks LANs and Wide AreaNetworks WANs attached together via a variety of transmission mediaaugmented by the Internet. These corporate communications worlds, i.e.,business enterprise's digital networks and the public switched telephonenetwork PSTN remain separate and distinct until now.

BRIEF DESCRIPTION OF THE DRAWINGS

The invention will be better understood from a reading of the followingdetailed description in which like reference numerals designate likeelements and to which:

FIG. 1 is a diagram of a Call Management System;

FIG. 2 is a block diagram of a call management computer;

FIG. 3 is a block diagram of a digital signal processor;

FIG. 4 is a detailed block diagram of a portion of the digital signalprocessor of FIG. 3;

FIG. 5 is a diagram showing how callers are identified;

FIGS. 6 a through 6 e show components of the call management window asthey appear at a workstation display;

FIG. 7 shows components of call management windows for VIP rule creationand management;

FIG. 8 shows a Fax handling display;

FIGS. 9 a and 9 b show components of call management windows; and

FIGS. 10 a and 10 b show “copper bypass” configurations for CallManagement System fault tolerance.

DETAILED DESCRIPTION

1. Overview

1.1 General

1.2 CO Trunks

1.3 PBX Trunks

1.4 DSP Processing & Switching

1.5 Digital Data Networks

1.6 Call Management Databases

1.7 Call Management Computer

1.8 Call Reception

1.9 Call Origination

1.10 “One Number” Processing

1.11 Called Party Identification

1.12 Call Type

1.13 Proactive Caller Identification

1.14 VIP Rules

1.15 Notification and Control via the Called Party's WorkstationComputer

1.16 Call Management

1.17 “Answer” a Call

1.18 “Transfer”

1.19 “Send to Voice Mail”

1.20 “Conference”

1.21 “Hold”

1.22 “Mute”

1.23 “Record” and “Playback”

1.24 “Hang Up”

1.25 “Outdial”

1.26 Calls Received for Non-System Users

1.27 Predefined Call Routing

1.28 Calls Originated by Non-System Users

1.29 Telephone Requirements

1.30 International CallBack

2. Voice Pathways

3. Real-Time Protocol and Signal Conversion

4. Intelligent Call Management Through Real-Time DSP Voice and DataProcessing and Circuit Switching

4.1 DSP Subsystem

4.2 Computer Signal Bus Interface

4.3 Dual-Port RAM

4.4 DSP Signal Processing Task

4.5 External Connectivity

4.6 DSP Motherboard

4.7 DSP Daughterboard Block Diagram

4.8 Trunk Interfaces

4.9 PBX Connections

4.10 Telephony Signal Buses

4.11 Circuit Switches

5. “One Number” Reception of Voice, Fax, Data Calls

6. Proactive Caller Identification

7. Continuously-Improving Caller Identification Databases

7.1 Calling Number Databases

7.2 Voice Name Identification

8. Call Notification & Control Via the Digital Network WorkstationComputer

8.1 Call Notification of the Called Party

8.2 Customer Logo

8.3 User Status

8.4 The Message Board

8.5 “FAX” Notification

8.6 “Flash” Mail Notification

8.7 “E-Mail” Notification

8.8 “Voice-Mail” Notification

8.9 User's Call Status

8.10 Call Alert Box

8.11 Workstation Real-Time Call Controls and Management

8.12 “Directory” Support

8.13 Call Origination

8.14 “Outside” Employee Support

8.15 “Group Secretary” Support for Calls to Specified Groups ofEmployees

8.16 “Meeting” Support for Users Away from Their Workstation

8.17 “Specialty-List” Support for Special Employee Groups

8.18 Feature Activation

8.19 TAPI Client

8.20 Automatic Updating

9. Multiple Call Handling Using a Single Extension

10. User-Defined VIP Call Handling

10.1 Temporary VIP Rule Usage

10.2 Advanced Message Notification

11. Routing Calls Inside or Outside the Organization

12. “Call Tags”

13. Facsimile Fax and Data Calls

13.1 Receiving Fax and Data Transmissions

13.2 “FAX” Notification

13.3 Unique Call Routing for Faxes or Data

13.4 Special Data Calls

13.5 Laptop Data Calls

13.6 Outgoing Fax and Data Transmissions

13.7 Retrieving Fax or Data Files Via “One-Call” Message Retrieval

14. User-Accessible Call Logs

15. “One-Call” Message Retrieval

16. Voice Mail Handling

16.1 Transferring Callers to Voice Mail

16.2 Alerting System Users to New Voice Mail Messages

16.3 Integrated Voice Mail Subsystem

17. User Status

18. Fault Tolerance and “Copper Bypass”

18.1 “Copper Bypass” Fault Tolerance

18.2 “Dual-System” Fault Tolerance

1. Overview

1.1 General

FIG. 1 is an overall block diagram of one embodiment of the improvedCall Management System, in which call control is provided by the userthrough a networked workstation computer; not a conventional telephoneinstrument. FIG. 1 shows the organization's environment with its LocalArea Network and/or Wide Area Network (LAN/WAN), an on-site system userwith a LAN/WAN based workstation, a PBX or similar switch, voice mailand a call management computer. In FIG. 1, an organization utilizing theCall Management System is clustered at the bottom of the figure and theoutside world of callers and system users is clustered at the top. Theorganization's calls are handled using a call management computer whichis placed so as to intercept telephone and data trunks between thetelephone provider's central office and the organization's PBX or otherswitch (or as its replacement). Also shown are a work-at-home systemuser with workstation connected via the Internet, a voice caller at apay phone as well as Fax and data callers all connecting through thepublic switched telephone network.

All central office calls ring directly into the Call Management System,the system has direct access to all information being provided by thecentral office. Since the system gets the “first look” at all incomingcalls, it can direct and process calls according to user requirements.

In the Call Management System of FIG. 1, call control is providedthrough a user workstation 114 to provide new and improved capabilitiesfor the user and substantially eliminating the shortcomings anddisadvantages of past systems.

FIG. 1 shows pictorially the public switched telephone network (PSTN)with voice 118, Fax 119, a call management system 99 coupled to thepublic switched telephone network 100 through a telephone central office(CO) 103. The call management system 99 includes a PBX or similar switch104 and connections to user telephone instruments 106. A digital datanetwork 109 attaches to user workstation computers 114 a-114 n. Thedigital data network of the illustrative embodiment is a conventionalLocal Area Network (LAN) or Wide Area Network (WAN). In FIG. 1, twodifferent types of system users are shown, the first is an in-house user113 associated with workstation 114 a and telephone instrument 106 awhile a second user is a work-at-home or traveling employee with aworkstation or laptop computer 114 p attached remotely via the Internetor WAN extension of the LAN 109, through ISDN or otherwise. The PSTNallows access to/from the call management system 101 via voicecommunication device 118, fax device 119, data 120.

A call management computer 101 is placed so as to attach to five 5separate interfaces described below.

1.2 CO Trunks

Call Management System 99 is coupled to central office 103 via CentralOffice trunks 102 for both voice and data connections.

CO trunks 102 includes a variety of trunks, including analog, DID, ISDN,T-1, DID over T-1, 800/900 T-1 services, data, and Internet. The centraloffice 103 is interconnected within the public switched telephonenetwork 100 via Local Exchange Carriers, Inter-exchange long-distanceCarriers, Cable companies, RF or satellite carriers, digital Internetproviders or any other types' of voice or data carriers. CO trunks 102may include multiple individual “circuits” ISDN, T-1, etc. which carryvoice and/or data for individual calls.

Voice calls over Internet and similar means are processed utilizingconventional digital techniques but are then fed into the system asthough they were voice trunks. For the purposes of this description, COtrunks include Internet connections.

Call management computer 101 is configured and programmed to appear totelephone service providers 103 as though it is a business PBX or otherbusiness telephone switch and/or an Internet or other data server ornode.

1.3 PBX Trunks

Within the call management system 99, PBX trunks 105 are the means bywhich the Call Management System 101 provides voice or data connectionsto system users or workstations or other devices within the business 99.For traveling or work-at-home users 111, access to the outside using COtrunks is considered just a part of PBX trunk access, e.g., voice callsto an at-home system user's 111 telephone 106 p or Internet voice ordata connections.

“PBX” includes a variety of different telephone switches includingclassical private branch exchanges PBXs, automatic call directors ACDs,key telephone sets, or integrated switches within the call managementcomputer 101. Telephone instrument 106 a-106 n, as shown as physicaltelephones but may also include headsets, earpieces, computer soundsystems, isochronous network technology such as isoEthernet and ATM, andother means of providing a voice or data connection to a user.

The call management computer 101 may attach to the organization's PBX orother telephone switch 104 through PBX trunks 105. The call managementcomputer can sit in front of virtually any type of switch. Noswitch-specific hardware or software is required for integration. PBXtrunks 105 may be analog, DID, DISA, ISDN, T-1, DID over T-1, 800/900T-1 services or other available types in all available variations andcombinations. In addition, the call management computer 101 may beconnected directly to the organization's telephone instruments 106 a-106n or directly to the user's workstation 114 a-114 n for voice or dataconnections in place of a switch 104.

PBX trunks may or may not be of the same kind and/or number as the COtrunks 102. The Call Management System may provide a one-to-one directrelationship between CO trunks 102 and PBX trunks 105 or it may provideprotocol “conversion” between differing CO and PBX trunk types and/ornumbers. PBX trunks also include direct connections to the user'stelephone instruments 106.

The call management computer 101 is so configured and programmed that itappears to the business PBX or other switch as though it is a centraloffice and/or it appears to the direct telephone instruments 106 a-106 nas though it is a business PBX switch such as 104.

1.4 DSP Processing & Switching

Trunk interfaces 203, 206, circuit switches 204 and DSP digital signalprocessors 208 interact with and control the CO and PBX trunks under theoverall control of the call management computer 101.

All CO trunks 102 and PBX trunks 105 are attached to the call managementsystem through appropriate trunk interfaces 203 and PBX trunk interfaces206. The interfaced trunk signals are further attached through circuitswitches 204 and high-speed telephony buses 210 to each other and tospecial DSP's 208.

The configuration of the call management computer 101 with individualinterface boards 203, circuit switches 204, DSP processors 208 and thehigh-speed buses 210 provide means for real-time sensing, switching andmanagement of calls and the means for the call management computer 101to appear to the central office 103, through the CO trunks 102, 202, asa business PBX 104 or other switch and/or a server computer for datafunctions. This configuration further permits computer 101 to appear tothe business PBX or other switch 104 or to the user's telephoneinstrument through the PBX trunks 105, 205 as a central office switchsuch as 103 and/or a data server.

1.5 Digital Data Networks

Notification of events and control over multiple calls is accomplishedindependently of the organization's PBX or other switch system even ifthe user's telephone instrument 106 a-106 n is currently busy becausethe digital networks 109 are separate from and independent of the user'stelephone instrument 106 or telephone system 104.

The call management computer 101 attaches to the organization's digitaldata network including a LAN as well as Internet and/or other externalWAN networks such as Internet via interfaces 209, 213 and 214, throughwhich it has immediate access to the user workstations, whetherin-the-office 113 or at the site of a remote user 111. These networksoperate independent of whether the user's telephone instrument 106 isbusy or not. The digital networks 109 are used by the call managementcomputer 101 to alert called users such as users 111 and 113 to incomingvoice calls and newly received Fax, voice or data messages and forreceiving back user controls of all types from the user's workstation114. In addition the digital networks 109 provides access to theorganization's LAN server computers 110 for e-mail, voice mail,database, Internet access and other services.

1.6 Call Management Databases

The call management system 101 utilizes a variety of interactive callmanagement databases 215 for functions including: system and userconfigurations, primary and secondary caller identifications and voicecaller identification, YIP rules, phone directories, Fax and data filestorage, voice message storage, user-accessible call logs and many otherfunctions. These on-line, real-time databases 215 may reside on the callmanagement computer 101 itself or elsewhere on the digital network,e.g., on a LAN-based database server 110.

The Call Management System structure of the embodiment includes callmanagement computer 101, the CO Trunks 102, the PBX Trunks 105, thetrunk interfaces 203, circuit switching 204 and DSP processing 208, theorganization's digital network(s) 109, 209 and the call managementdatabases 215. However, implementation may be in various combined orextended ways, such as when the call management computer 101 is builtinto the central office 103 or into a PBX or similar switch 104. Thecall management computer 101 may replace the PBX and control theorganization's telephone instruments 106 directly.

1.7 Call Management Computer

The call management computer 101 is configured as shown in FIG. 2. It isbased on an industry-standard computer with processors, memory, powersupply and cabinetry. The computer 101 is coupled to a data bus 211. Thedata bus has connections to LAN interface 209 and disk memory whichstores Call Management Databases 215. The databases may alternativelyreside on the digital network system 109. The data bus 211 is connectedto interfaces for Digital Internet connections 213 and bulk calling lineidentification BCLID data link 214 to the central office.

The telephony subsystem ties together the CO trunk 102 and PBX trunks105 through their specific trunk interfaces 203, 206 and circuitswitches 204 to the telephony signal buses 210. Each of the trunkinterfaces 203, 206 is also coupled to the computer data bus 211,through which the computer processors 201 receive information andprovide control and data to both the CO trunk interface 203 and itscircuit switches 204. The DSP digital signal processors 208 includemultiple DSPs. The multiple DSPs as needed are attached to the telephonysignal buses 210 through switches 204 and to the computer data bus 211through which they provide information to the computer processor 201 andreceive back control and data e.g. voice messages to play out to thecaller. For voice-over-Internet or similar digital connections, voiceinterface board 207 is connected to the computer data bus 211 andthrough its own circuit switches 204 to the telephony signal bus 210,through which the voice connections can be made to/from any telephoneinstrument 106.

1.8 Call Reception

Typical call paths 121, 221 are shown on FIGS. 1 and 2. The in-band callinformation from central office 103 is sent through a CO trunk 102 tothe call management computer 101 where it attaches through anappropriate CO trunk interface 203 and circuit switch 204 to thetelephony signal buses 210. For each trunk/circuit, the call managementsystem assigns one or more DSP processors 208 connected to the telephonysignal bus 210, to provide a monitoring and control link 219 for thatcall.

An incoming call such as from the payphone caller 118 or the Fax caller119 or data callers 120 is routed through the PSTN 100 to the centraloffice 103 and then to the call management system 99 through a CO trunk102. The assigned DSP 208 and/or CO interfaces 203 monitor for anincoming call analog or digital signals in any available form,appropriate to the type of trunk and/or circuit.

When the call is presented by the CO 103, the call setup commands arerecognized through the trunk interface 203 or through the associated DSP208 and the call management computer processors 201 receives thisinformation via the trunk interface 203 or DSP 208 connections to thecomputer signal buses 211. Control signals from the call managementcomputer 101 then cause the call to be answered via the same routeaccording to the trunk and circuit type.

At this point in the process, a first call path segment 221 a to a DSPmonitoring and control link 219 for the trunk and circuit terminates atthe assigned DSP 208.

Connections to a system user 113 are created by the call managementcomputer 101 selecting an available, appropriate CO trunk inbound 105and establishing a call to the PBX 104 or to remote system users overadditional trunk 102 to the central office 103. The PBX 104 or CO 103responds to the call setup commands depending upon the type of trunk andcircuit (including voice-over-Internet and other digital services).These are sensed by the trunk interface 203 or 206 or the assigned DSP208 and passed to the call management computer 101. Call managementcomputer 101 then controls the appropriate circuit switches 204 toconnect the voice pathway from the calling party 118 to the voicepathway to the called party's 113 telephone instrument 106 or via asecond call path segment which includes segment 218 and call pathsegment 221 b, leaving the assigned DSP 208 attached to continueproviding the DSP monitoring and control link 219.

The call is then put through in a conventional manner by the PBX 104 orCO 103 to the called party's 113 or 111 telephone instrument 106 a or106 p where it rings and is answered by the called party 113 or 111completing the connection between the caller 118 and the called party113, 111. The typical call path 221 with the associated DSP monitoringand control link 219 is then completed as shown. The voice pathway socreated may be reused as described below.

The CO trunk interfaces 203 and PBX trunk interfaces 200 and assignedDSP 208 remain active throughout a call or series of calls to adestination, watching for either end to terminate the call, by hangingup the telephone instrument, or otherwise changing the call state whilethe call management computer 101 watches for the system users 111, 113to select a command changing the call's state.

1.9 Call Origination

Calls are originated by system users through their workstation 114. Suchoriginated calls may be destined to any system user 113, 111, anon-system user, or anywhere else in the PSTN 100. Depending upon thedestination, the call management computer 101 selects an available,appropriate CO trunk 102 or PBX trunk 105 and establishes the call tothe CO 103 or to the PBX 104 or the telephone instrument 106 usingappropriate signaling techniques for that trunk or circuit. The callmanagement computer 101 then instructs the circuit switches 204 toconnect the call circuits together, “bridging” the originator to thedestination and creating the typical call path 118 in FIG. 1 and DSPmonitoring and control link 119 as shown in FIG. 2.

The CO trunk interface 203, the PBX trunk interface 206 and assigned DSP208 remain active throughout each call, monitoring for either end toterminate the call, by hanging up the telephone instrument, or otherwiseto change the call state while the call management computer 101 monitorsfor the system user 111, 113 to select a command changing the call'sstate.

The call management computer 101 manages the available, appropriate COand PBX trunks 102, 202, 105, 205 so as to share between non-system usercalls established from the PBX or telephone instruments and system user113, 111 calls established by itself.

The call management computer software runs under Microsoft's Windows NToperating system, a multi-threading, multi-tasking operating systemrequired by the real-time call management aspects of the system. TheCall Management System may be configured as a “client” to theorganization's existing digital networks or as a “server” whenincorporating Internet or other server functions for the digitalnetwork(s).

1.10 “One Number” Processing

For system users 111, 113, the Call Management System uses only a singleDID (direct in dial) or extension number to receive all calls, voice,fax or data in any mixture and number within the limit of the number ofavailable trunks and circuits at any time or all at the same time. Thecall management computer 101 is programmed to sort them out and handleeach appropriately.

1.11 Called Party Identification

After a call is received, the call management computer 101 determines adestination party for the call, either automatically through receptionof DID, DNIS (dialed number identification service), ISDN or othersignals or messages from the central office 103 as detected by the trunkinterface 203 or the assigned DSP 208 or, alternatively, a callattendant feature of the Call Management System.

In Call Attendant mode, the call management computer 101 instructs theDSP 208 to play out one or more voice messages from the call managementdatabase 215 asking the caller 118 to identify the destination party byname, spelling, extension number or otherwise. The DSP 208 receives theinformation from the calling party 118 and passes it to the callmanagement computer 101 where the called party 111 or 113 is identifiedthrough the digits entered, through voice recognition or otherwise. Thecalled party's extension number is found or verified using the callmanagement database.

For voice over Internet or similar techniques, the caller is providedand fills in a “form” which includes: the name of the caller, the nameof the called party and other appropriate information.

1.12 Call Type

The incoming call type voice, Fax or data is also determined using theDSP 208, which searches for appropriate signaling from the call source118 e.g. none for voice, or CNG for Fax, specified DTMF, carrier orother signals for data whether files, video data, video conferencing,etc. in analog or digital form.

For non-voice calls to system users, the attached DSP 208 is instructedto switch to the appropriate Fax or data mode and to receive thetransmission automatically for storage in the call management database215 and later use by the called party 111, 113 or to transfer the callautomatically to an appropriate extension, e.g., for video conferencing.

1.13 Proactive Caller Identification

The identity of the voice caller 118 is determined either automaticallythrough reception of Caller ID, ISDN, ANI, BCLID or other informationfrom the central office 103 as received by the trunk interface 203, theassigned DSP 208, the BCLID data link 214 or otherwise. Depending uponwhat information was received, if any, Proactive Caller Identificationmay then use direct interaction with the caller 118 and the Caller IDdatabases 215 for additional information. Proactive CallerIdentification is described in Section 6 and the Caller ID databases aredescribed in Section 7.

1.14 VIP Rules

Specific rules, called “VIP rules”, are created to specify specialhandling for important callers, sets of callers or even for all callers.These VIP rules precede and augment direct user controls and aredescribed in Section 10.

1.15 Notification and Control Via the Called Party's WorkstationComputer

Voice calls destined for system users either working in the office 113or outside the office 111 are handled by alerting the called party athis workstation 114 through sending a message from the call managementcomputer 101 across the digital network 109 to the user's workstation114 and thus to the user's call management window 115 which “pops up”onto the user's workstation screen. The user then controls the callthrough selections made using his call management window 115 and itssubscreens. This procedure, the windows and the user's functions aredescribed in Section 8.

1.16 Call Management

Once the call type, called party and caller are identified, the callmanagement computer 101, 201 handles these calls based first on anyapplicable VIP rules and then on commands from the user, via theirworkstation call management window 115. A call is “held” and manipulatedby the call management computer 101 throughout answering of a call,identification of the call type, identification of the called party,proactive identification of a calling party, playing out messages to thecaller, receiving information received from the caller, VIP rulehandling and/or notification of the called party 113. The typical callpathway 221, 219 is terminated at the assigned DSP 208 and is not passedbeyond.

Some of the call functions which may be exercised by a system user aredescribed in the following paragraphs insofar as what happens with thecall management computer 101. Detailed descriptions of the callmanagement window 115 and the user's functions, controls and managementusing their workstation 114 are provided in Section 8.

1.17 “Answer” a Call

A system user such as user 111 or 113, may select the “Answer” functionfor a call announcement. If the user has no calls currently active, thecall management computer 101 selects an available CO trunk (inboundtrunk 105 for user 113 or outbound trunk 102, for user 111) andestablishes a call to the PBX 104 or the central office 103. The PBX 104or central office 103 responds to call setup commands depending upon thetype of trunk and circuit. Call set up signaling is sensed by the trunkinterface 203 or 206 or an assigned DSP 208 and passed to the callmanagement computer 101 which then controls the appropriate circuitswitches 204 to connect the voice pathway from the calling party 118 tothe voice pathway 121 to the called party's 113 or 111 telephoneinstrument 106 a or 106 p, leaving the assigned DSP 208 attached as wellto continue providing the DSP monitoring and control link 219.

The call is then processed conventionally by the PBX or CO to connect tothe called party's 113 or 111 telephone instrument 106 a or 106 b whereit rings and is answered by the called party 113 or 111 to complete theconnection between the caller 118 and the called party 113 or 111. Thetypical call path 221 with the associated DSP monitoring and controllink 219 is then completed as shown. The voice pathways so created maybe reused as described in Section 2.

For cases where the Call Management System is connected in the centraloffice 103 and replaces the PBX and controls the telephone instrumentsdirectly, the process is basically the same.

If a system user such as user 113 or user 111 currently has an activecall, the voice pathway 121 already exists and the user's current callwill be moved to “hold” or “hang Up” mode as defined by the user and thevoice pathway switched immediately to the new caller (see “Transfer”below).

The CO trunk interfaces 203 and PBX trunk interfaces 206 and assignedDSP 208 remain active throughout a call or series of calls to adestination, monitoring for termination of the call by either endthrough hanging up the telephone instrument or otherwise changing thecall state. Call management computer 101 monitors for the system users111 or 113 to identify selection of a command from the user's callmanagement window 115 to change the call's state.

1.18 “Transfer”

The “Transfer” function is used to cause a received call to betransferred to another destination either inside the business 99 or toany destination coupled to the PSTN 100. Transferring a call requiresthe user to select a “Speed-Transfer” button or a “Transfer” screen fromwhich he may select a destination from a directory or the user may typein the phone number to use for the destination.

To transfer a call to another destination, the call management computer101 receives a transfer message from a user's call management window 115via the digital networks 109, the call management computer 101 instructsswitches 204 to disconnect the voice path 121, instructs the appropriatetrunk interface 206 to “hang up” the call to the user if appropriate andinstructs the DSP 208 to return to call monitoring.

If the new destination is a system user, call management computer 101checks for any appropriate VIP rules for this calling party and thecalled party and processes the call as specified by the VP rules.Otherwise, call management computer 101 alerts the called party andawaits user control.

If the destination is not to another system user, the call managementcomputer 101 establishes a new voice pathway to the destination whereverit may be, instructs the appropriate switches 204 to connect the voicepathways together 221 and controls the trunk interface 103, 106 and theDSP 108 to monitor the progress of the call, searching for hang up orchange of state at either end.

1.19 “Send to Voice Mail”

One special function provided by the call management window 115 is tosend a call to voice mail whether the call is active or not. This is aspecial variant of the “Transfer” function with a pre-specifieddestination.

1.20 “Conference”

The Call Management System can conference calls independent of whetherthe parties are directly coupled to the PBX or are accessed via thecentral office 103. When a system user 113 selects the “Conference”function for an existing active call or with no active call, he thenselects one or more destination parties from the call management window115 directory and/or types in one or more telephone numbers. The callmanagement window 115 sends a “Conference” message down theorganization's digital network(s) 109 to the call management computer101, 201, whereupon the call management computer 101 alerts the newsystem users.

As each called system user “Answers” the Conference call, the callmanagement computer 101, 201 creates a new voice path 121, asappropriate, instructs the appropriate circuit switches 204 to connectthe voice paths together to the assigned DSP 208, and instructs the DSPsto combine the signals appropriately to create a conference call.

For “Conference” destinations which are not system users, the callmanagement computer 101, 201 immediately establishes a call to thedestination and “bridges” the circuits, as with “Outdial” callsdescribed below.

1.21 “Hold”

A system user can select the “Hold” function for any call currentlyactive. The “Hold” messages are sent by the call management window 115via the organization's LAN or WAN 109 to the call management computer101, call management computer 101 instructs the appropriate circuitswitches 204 to break both the inbound and outbound portions of thevoice path 121 effectively placing a caller on “hold” but providing theuser rapid access to the call through simple mouse clicks. During unusedtime when the calling party would otherwise be waiting, the CallManagement System provides “Music on Hold” and/or corporate messages(sales, information, etc.). The caller can terminate any messages at anytime by entering a “#”.

1.22 “Mute”

A system user can select the “Mute” function for any call currentlyactive. “Mute” messages are relayed via the call management window 115and sent down the organization's LAN or WAN 109 to the call managementcomputer 101 which instructs the appropriate circuit switches 204 tobreak just the outbound portion of the voice path from the system userto the caller leaving the inbound portion active, thus muting the call.

1.23 “Record” and “Playback”

A system user can select the “Record” function for any call currentlyactive. The “Record” messages are relayed via the call management window115 and sent via the organization's digital network(s) 109 to the callmanagement computer 101 which instructs the assigned DSP 208 to recordthe call content, sending it to the call management computer 101 whichsaves it to the Call Management Databases 215 for future replay.

1.24 “Hang Up”

When either calling or called party terminates a call by hanging up orby selecting the “Hang Up” function from his call management window 215,the call path 121 is dismantled by the call management computer 101instructing the switches 204 to disconnect the voice path, instructingthe appropriate trunk interface 203 or 206 and/or DSP 208 to “hang up”the call as appropriate and instructing the DSP to return to searchingfor a new call. However, either end currently with new calls waiting iskept active as a re-usable voice pathway.

1.25 “Outdial”

The users of the system 99 may originate calls to other users located atbusiness 99 or to any numbers outside the business 99. The user utilizeshis/her call control window 115 to identify or type in a destination forthe call internal or external and then instructs the Call ManagementSystem to dial to that destination.

Workstation 114 sends the dialing control messages via the digitalnetwork(s) 109 to the call management computer 101, 201 which, in turn,causes the call to be placed using an available CO or PBX trunk/circuit102, 105 of the appropriate type through the circuit switches 204. Whena call is in process or completed to the destination, a voice path 121is established to the calling party's telephone instrument 106.

Once established, the system user 11, 113 has available all of thesystem features for originated calls, as for inbound calls.

1.26 Calls Received for Non-System Users

Calls received for business 99 employees who do not have appropriateworkstations or who do not choose to be system users, or for theorganization's voice operator and for dedicated Fax machines and otherhardware devices are immediately switched by the call managementcomputer 101 to the PBX 104 or telephone number dialed, and are handledconventionally. For such calls, the call management computer 101, 201selects an available and appropriate PBX trunk 105, establishes a callto the desired extension controls and the circuit switches 204 toconnect the voice paths together. DSP 208 monitors progress of the calland monitors for either end to hang up. This process includes protocolconversion features for CO trunk/circuit 102, which are of differenttype than the PBX trunk/circuit 105.

1.27 Predefined Call Routing

Fax or data calls received for specified numbers, such as the business'sbase number, are accepted as though directed to a specified user, e.g.,the organization's operator, who is then expected to sort out and theFaxes send them to the appropriate users or print them.

1.28 Calls Originated by Non-System Users

Calls originated by non-system users, via their PBX or through theirtelephone instruments controlled directly by the Call Management System,are handled conventionally except that, the call management computer 101receives the call establishment from the PBX 104 through the PBXtrunk/circuit 105 and passes it on to the central office 99 via anavailable CO trunk circuit 102, connecting the circuits together viaappropriate circuit switches 204. The user is unaware of this processand sees no difference from conventional telephone usage.

1.29 Telephone Requirements

One significant advantage of the Call Management System is that itprovides system users the many unique features and functions describedhere while requiring nothing more than plain old telephone services(POTS) telephones or headsets instead of expensive multi-buttonproprietary business telephone instruments. None of these featuresrequire the telephone instrument to be anything more than just a voicepath to the user's ear and mouth.

1.30 International CallBack

The Call Management System provides a “CallBack” subsystem with which acalling party places a call from a foreign telephone, the systemreceives the call and telephone number, then terminates the call andimmediately dials the caller back at the number received. This processsaves significant telephone expenses compared with the costs of callsfrom many foreign countries.

2. Voice Pathways

When a call is put through to a system user, the call managementcomputer 101 creates a reusable “voice pathway” 121 to the called partyeither an in-house user 113 or an external user 111. Voice pathwayswitching, rather than establishing and tearing down multiple separatecalls, provides the ability to switch rapidly between multiple calls, ondemand, based on the user's changing priorities through simplepoint-and-click mouse, keyboard or menu operations.

FIG. 1 shows two re-usable voice pathways 121, one to the “in-house”system user 113 via PBX trunks 105 and the organization's PBX 104 to theuser's telephone instrument 106 and another to the “at-home ortraveling” system user 111 via a CO trunk/circuit 102 to the centraloffice 103, and then via the PSTN 100 ultimately to the user's telephoneinstrument 106. Both of these are valid cases, even though they utilizedifferent CO and PBX trunks of any appropriate type. Note that the CallManagement System does not care whether the called party is at anin-house extension or is not directly connected to the call managementsystem but is anywhere reachable via the PSTN or through voice overInternet or some other network.

A voice pathway 121 is created by the call management computer 101 inany of a variety of ways:

1. Placing a call through the PBX trunks 105 and the organization's PBX104 or other switch to a called party's extension 106;

2. Activating directly a called party's telephone in the case where thecall management computer 101 directly controls the telephone instruments106;

3. Dialing out through the PSTN to a remote called party 111;

4. Connecting to a called party using an external network such as theInternet;

5. Connecting through remote links to another of the organization's PBXor other switches;

6. Passing voice over the organization's communications infrastructurewithin the site or external to it;

7. Playback of recorded voice messages may be done via a “Voice Pathway”created to the user's telephone instrument 106 or via the “Data Path”transferring the voice message to the user's workstation to be playedout via the workstation's sound capability.

8. Any other such means which can establish a voice connection to auser's telephone instrument as defined.

Once established, the voice pathway 121 is used for the entire durationof that call and all other calls dialed or transferred to that samedestination, until all such calls have been processed and the voicepathway is no longer needed.

Once a voice pathway is established to a system user such as user 113 oruser 111, the calls being held in the call management computer 101 forthat user may be rapidly “switched” 204 to a destination voice pathway121 on demand as controlled by the user 113 or via the user workstation114 through the call management window 115.

The central office trunk interfaces 203, PBX trunk interfaces 206 andassigned DSP 208 remain active throughout a call or series of calls to adestination, monitoring for either calling or called party to terminatethe call or otherwise change the call state. Likewise the callmanagement computer 101 monitors for the system user 113 to requestchange of the call state, e.g., “hangup”, “Hold”, etc. When appropriate,the call path is dismantled by instructions to the circuit switches 204and DSP 208 from the call management computer 101.

If either calling or called party has other calls waiting, the callmanagement computer 101 does not “hang up” that end of the call path121. Instead it is kept so that it may be re-used as a voice pathway forthe waiting calls. Otherwise, the call management computer 101 instructsthe trunk interface 203, 206 and/or DSP 208 to “hang up” the call andreturn to waiting for another call to be presented.

Voice pathway switching, rather than requiring the establishment andtearing down of multiple separate calls as conventionally done, providesthe ability to switch rapidly between multiple calls, on demand, basedon the user's changing priorities.

3. Real-Time Protocol and Signal Conversion

The Call Management System provides for real-time protocol conversionbetween central office trunk type 102 and number and PBX trunk/circuit105 types and number. The number and type of central officetrunk/circuits need bear no direct relationship with the number and typeof PBX trunk/circuits. This conversion allows the Call ManagementSystem's new and expanded features and functions to be implemented usingexisting telephone systems which cannot otherwise accept new telephonecapabilities directly or in a cost-effective manner.

As shown in FIGS. 1 and 2, the call management computer 101 attaches tocentral office trunk/circuits 102 from one or more central offices 103on the one side and through PBX trunk circuits 105 to the PBX 104 and/ordirectly to telephone instruments 106. Each type of CO trunk and/or PBXtrunk analog, analog DID, T-1, DID over T-1, ISDN, Internet or others isattached through its own appropriate type of interface board 203, 206which converts the trunk signals to standardized bus signals for thecircuit switches 204 and telephony signal buses 210 and it also monitorsvarious aspects of the call.

To identify both the calling party 118, 119, 120 and called party 111,113 as automatically as possible, it would be logical for theorganization to elect to subscribe to modern telephone provider serviceseven though the organization's existing PBX cannot reasonably or costeffectively be upgraded to support such new services. One example ofsuch new services would be ISDN PRI, where the digital “D” channelidentifies the telephone number of both the calling party and the calledparty for each of up to 23 or 24 circuits. ISDN PRI provides a goodstart toward automating proactive caller identification.

While most central offices 103 can provide ISDN PRI or other new COtrunk/circuit 102, 202 services, existing PBXs 104 are customarilyoutfitted with old-style analog DID or even simple analog 1 FB or othertrunks having entirely different signaling and control requirements fromthe new ISDN PRI. Rather than requiring an expensive or even impossibleupgrade of the existing PBX 104, the call management computer 101provides “Conversion” of the new types of CO trunk/circuits 102 withtheir new signaling requirements to older PBX trunk/circuits 105 withtheir older signaling requirements.

New ISDN PRI services are provided by the central office 103 through theCO trunks 102 to the call management computer 101, 201 where thecircuits are “converted” to older analog DID services for the PBX trunks105 to the organization's existing PBX adding all of the Call ManagementSystem features and functions without changing the organization'sexisting PBX investment.

“Conversion” is made possible by the structure of the call managementcomputer 101 with the logical and physical separation between trunks,where each different CO trunk 102 and PBX trunk 105 has its own uniquetrunk interface 203 and 206 which converts the unique trunk signals tostandardized circuit signals for the circuit switches 204 and telephonysignal bus 210. The only thing that is common is that their voice pathscan be connected together, when appropriate, using the circuit switches204 and telephony signal buses 210. The handling of trunks/circuits fromone side is independent of and separate from the handling oftrunks/circuits from the other, yet their voice paths can be connectedtogether to complete the circuit whenever appropriate, coming in via onetype of trunk interface, converting to standard signals and going outvia an entirely different trunk interface being “Converted” en route.

Because of this “Conversion” ability, no direct one-to-one relationshipneed exist between the CO trunk/circuits 102 and the PBX trunk/circuits105. A special case occurs where the call management computer 101directly connects to and controls the organization's telephoneinstruments 206 with no PBX switch 204 at all

Real-time protocol and signal conversion thus provides a crucialenabling mechanism for all of the new features and functions of the CallManagement System using information available through new types oftelephone provider services, while still using an existing PBX whichcannot otherwise support such services. In effect, the Call ManagementSystem only requires the legacy PBX to provide voice paths to usertelephone instruments, to voice mail and other devices ignoring itsother existing features.

4. Intelligent Call Management Through Real-Time DSP Voice and DataProcessing and Circuit Switching

The Call Management System's unique “Intelligent” call managementcapabilities are based upon the real-time sensing, control, voice anddata processing provided by the call management computer's 101configuration of DSP processors 208, central office trunk interfaces203, PBX trunk interfaces 206 and circuit switches 204 through whichcircuits calls are assigned to one or more DSP processors 208 at alltimes. The DSPs provide real-time DSP voice and data processing which isthe essential means by which the content of each call is monitored andknown. The Call Management System monitors call content and uses thatinformation to provide “intelligent” call management, unlikeconventional PBXs or other telephone switches, which strictly avoidknowledge of the call content and are conceptually limited to “callswitching”.

The amount of DSP processing power applied up to at least 800 MIPS persystem coupled with the hardware sensors of the trunk interfaces 203allowing the Call Management System to apply a broad range of voice anddata processing tasks to each and every call, processing capabilitiesand power not available in conventional business PBXs or similarswitches.

4.1 DSP Subsystem

Each DSP subsystem consists of a DSP motherboard which attaches to thecall management computer 101, 201 via the computer signal buses 211 andto the telephony signal buses 210 through circuit switches 204. Inaddition, the motherboard supports one to four DSP daughterboards ofthree DSP processors each 3, 6, 9 or 12 DSPs per subsystem with 100,200, 300 or 400 MIPS of processing power.

Each call management computer 101, 201 utilizes one or more such DSPsubsystems.

The commercially available Analog Devices DSP ADSP-2181 is used for theDSPs, operating at 16.67 Mhz 33 MIPS with 32K bytes of program RAM and32K bytes of data RAM. DSP daughterboards are populated on the DSPmotherboard as required to provide system support for broad categoriesof services:

“Call Monitoring” DSPs are those assigned to support the actual numberof existing CO and PBX trunks, monitoring and controlling individualcalls e.g. call progress monitoring, voice playback, etc. Each callmonitoring DSP in this implementation typically can handle four throughcircuits/calls both CO and PBX sides for a total of 12 to 48 pairs perDSP subsystem.

“Special function” DSPs provide special functions as required for eachspecific system e.g. voice recognition, text-to-speech, etc.

Because of the anything-to-anything circuit switching capabilities ofthe Call Management System, DSPs can be assigned to any circuit/call asneeded to support any particular function or set of functions.

4.2 Computer Signal Bus Interface

The DSP motherboard connects to the call management computer 101 throughits interfaces with the computer signal buses 211 through which the callmanagement computer's software drivers control, monitor and passinformation between the call management computer processors and memory201, the call management databases 215, the DSP motherboards and the DSPprocessors 208 on daughterboards.

The DSP motherboard uses an industry standard PCI bus interface withindustry-standard “plug-and-play” support:

1. Assignable interrupt level

2. Assignable shared memory addressing

3. Assignable control addressing

4. Other fixed and/or dynamically changeable parameters.

4.3 Dual-Port RAM

Each DSP has its own 32K bytes of dual-port shared RAM memory which itshares with the call management computer processors 201. Thus, eachdaughterboard uses 96K bytes of shared memory and a fully-populated DSPsubsystem uses 384K bytes of shared memory. The shared memory providescommunications between the DSP and the call management computer's 101,201 software drivers and minimizes the overhead of system interruptionsexternal memory delay states. Shared memory is used for:

1. Passing voice buffers for voice playback and record

2. Passing Fax and data buffers

3. Providing “mailboxes” for control and status information.

4.4 DSP Signal Processing Task

Each DSP operates with its own Multi-tasking software environment,sharing the available time and MIPS among a series of tasks. The numberand choice of which tasks are active at any moment depends upon thestate of each of several calls the DSP is handling. Each call is itselfconsidered a “state” machine. These tasks include:

1. DTMF decoding Dual Tone Multi Frequency, with talk-off play-offprevention

2. DTMF generation

3. MF decoding and generation

4. Call Progress decoding Dial Tone, Ring back, Busy, Fast Busy based oncadence, etc.

5. Precision Call Progress Decoding based on frequency

6. Call Progress tone generation Dial tone, Ring back, Busy

7. Analog signaling control FXO LS/GS, FXS LS/GS, E&M, E&M Wink, DPO,DPT

8. Caller ID decoding FSK modem signaling

9. Caller ID generation FSK modem signaling

10. Voice Playback messages played out to the caller

11. Voice Recording voice name for caller identification, recording ofvoice messages and recording of calls

12. Voice compression and decompression PCM, ADPCM, etc.

13. Conferencing of calls chimes and voice

14. CNG Fax tone detection

15. Fax Group 3 reception and transmission including CNG tone generationand special Fax identification for reception and special Fax banner fortransmission for each system user

16. X.36, X.34 and other modem protocols

17. ECLID data reception

18. Text-to-speech conversion

19. Text-to-Fax conversion

20. Voice recognition

21. Name recognition voiceprint

22. Many other tasks.

Most of these DSP signal processing tasks are not typically available inconventional business PBXs or similar switches but may available ininteractive voice response and similar equipment. In the future, manymore voice and data processing tasks will be added to the CallManagement System's DSP processors 208, significantly expanding thefunctions and features of the Call Management System.

4.5 External Connectivity

Each DSP motherboard also supports the following external connections:

1. External audio connection for “music-on-hold”

2. External audio microphone input which can be used for voice recordingif desired

3. External headphone connection so that calls in progress can bemonitored for debug purposes

4. Status LEDs red and green for self-test, board diagnostics andoperational external notification

5. “Stay-alive” signal for the “copper bypass” unit.

4.6 DSP Motherboard

FIG. 3 shows the DSP motherboard block diagram. The DSP motherboardfunctions primarily as an interface between the computer signal buses211, the telephony signal buses 210, the DSP daughter boards and theirDSPs 308 and internal decoders, sequencers and logic. This architectureutilizes common internal address and controls 315 and data 316 signalsfrom the PCI interface 301 connecting the call management computer's 201PCI bus 211 to essentially everything on the motherboard 208 and throughit to the DSP daughterboards. Each motherboard 208 includes:

1. The FMIC telephony bus circuit switches 204 which provide signalpaths to/from the telephony signal buses 210 and the DSP processors 208

2. The address counters 317, chip select circuits 319 and address andmemory sequencer 318 which manages the on-board controls

3. DSP and FMIC controls are handled via DSP selects 320, interruptcontrol register 321 and the mailbox and error register controls 322

4. External audio input for music on hold and headset for debugging areprovided 225 by a separate audio circuit.

Each daughterboard interface consists of:

1. Address bus 315

2. Data bus 316

3. Telephony voice circuits 313

4. Status and error signals to the DSP motherboard buffers and logic314.

4.7 DSP Daughterboard Block Diagram

FIG. 4 shows the DSP daughterboard. Each daughterboard 2081, 2082, 2083,and 2084 attaches to the DSP motherboard 208 through the daughterboardinterfaces address lines 315, data lines 316, telephony circuit buses313 and status/error lines 314. Each DSP 208, 308, 408 has its owndual-port RAM data memory 410 accessible either from the DSP or from thecall management computer 101, 201 via the PCI motherboard interface 301.

4.8 Trunk Interfaces

The call management computer's 101, central office trunk interfaces 203support one or more trunks per board, depending upon the type of trunks,with varying number of circuits per trunk typically 1 or 2 T-1 or ISDNPRI to 24 analog, DID or stations. The trunk interfaces 203, 206 providea variety of different connections, interfaces and features, as isappropriate for each different type, including the following centraloffice and PBX connections:

A. Central Office Connections

-   -   1. Loopstart—Ringing detection    -   2. Loopstart—Loop current detection    -   3. Loopstart—Loop current direction    -   4. Loopstart—On/Off hook control    -   5. Loopstart—Hookflash generation    -   6. Groundstart—Ringing detection    -   7. Groundstart—Loop current detection    -   8. Groundstart—Loop current direction    -   9. Groundstart—On/Off hook Control    -   10. Groundstart—Hookflash generation    -   11. Groundstart—Ring ground seizing trunk for outdialing    -   12. Groundstart—Tip ground detection of seized trunk    -   13. AnalogDID—Battery power to trunks    -   14. AnalogDID—Loop current detection    -   15. AnalogDID—Battery reversal wink generation    -   16. AnalogDID—Reverse battery answer supervision identifying        when the call is answered and when the station side disconnects    -   17. AnalogDID—DID decoding of DTMF signals    -   18. T-1 Trunks—Signaling type E&M Wink and Immediate, loopstart,        groundstart    -   19. T-1 Trunks—Wink generation    -   20. T-1 Trunks—“robbed-bit” signal decoding    -   21. ISDN PRI—“D” channel digital support        4.9 PBX Connections

1. Loopstart—Battery power to trunk

2. Loopstart—Ringing generation

3. Loopstart—Loop current detection

4. Groundstart—Battery power to trunk

5. Groundstart—Ringing generation

6. Groundstart—Loop current detection

7. Groundstart—Loop “open” for idle state

8. Groundstart—Tip Ground seizing trunk for outdialing

9. Groundstart—Ring Ground Detection of PBX initiating call

10. AnalogDID—Loop current detection

11. AnalogDID—Loop current direction detection for wink and answersupervision

12. AnalogDID—On/Off hook control

13. AnalogDID—DID transmission via DTMF

14. T-1 Trunks—Signaling type E&M Wink and Immediate, loopstart,groundstart

15. T-1 Trunks—Wink generation

16. T-1 Trunks—“robbed-bit” signal decoding

17. ISDN PRI—“D” channel digital message support

4.10 Telephony Signal Buses

In this implementation, the Telephony signal buses are based on theindustry standard Multi-Vendor Integration Protocol (MVIP) which providefor 256 bi-directional voice/data channels, divided into 16uni-directional or 8 bi-directional “streams” of 32 time slots each,operating at 2.048 Mhz. However, the buses could just as well have beenbased on SCSA, PEB or other types of available standards orconventions—so long as “clear”, full-bandwidth circuit paths areprovided among the trunk interfaces 203, 206, 207, the DSPs 208, andtheir circuit switches 204.

4.11 Circuit Switches

Trunk interfaces 203, 206, the Internet voice interface 207, DSP 208 andothers boards connect to the telephony signal buses 211 through theirown circuit switches 214. The circuit switches are the FMIC FlexibleMVIP Interface Circuit. The FMIC connects the specified time slots ofthe telephony signal buses 210 to/from the “on-board” internalcircuitry.

At any time, the call management computer's software drivers can changethe circuit switch 204 settings using commands through the computersignal buses 211. This is the means through which calls/circuits/voicepaths are dynamically “switched” from one point to another, placed on“hold”, or otherwise as described elsewhere.

5. One Number” Reception of Voice, Fax, Data Calls

Without changing existing information and telephone systems, the CallManagement System provides a single, unique “One Number” for each systemuser which is his “personal point-of-contact” and “never busy” telephoneextension (his Telco DID or equivalent number or his call attendant orDISA extension number) for all voice, FAX and data communications. This“one number” is used to receive, identify and automatically handle allthe user's voice, fax and data calls, one or several at a time usingmultiple trunk circuits, even when the user is on his/her telephone.Thus, user telephone extensions are converted into a “never busy”extension for voice, e-mail, FAX and data with direct control from thedesktop computer where users exercise direct, real-time control of allcalls including call queuing with multiple calls on hold, calltransfers, call forwarding and other forms of real-time call processingnot currently available. The use of only one number per usersignificantly reduces the costs, complexity, inefficiency and confusionof having multiple different telephone numbers for different functions.

The Call Management System uses only a single DID or extension numberfor each user such as user 111 or user 113 to receive all their directcalls, voice, fax or data in any mixture and number within the limit ofthe number of available trunks and circuits. Various calls to a systemuser may occur at any time or several may occur at the same time.

The Call Management System uses only a single DID or extension numberfor each user 111 or 113 to receive all direct calls, voice, fax or datain any mixture and number within the limit of the number of availabletrunks and circuits. Various calls to a system user may occur at anytime or several may occur at the same time. The call management computer101 is programmed to sort them out handle each appropriately, all at thesame time. This “One Number” feature is a significant improvement overthe conventional use of multiple numbers for different functions.

FIG. 1 shows four different types of callers: an outside voice caller118, an inside voice caller 113, an outside Fax caller 119 and anoutside data caller 120. These callers all use the same “One Number” tocall the same system user 111 and they may do so all at the same time.

So long as a sufficient number of trunks or trunk circuits areavailable, all of the outside calls will be routed through the PSTN 100to the CO 103 where they will be sent to the organization via the COtrunks 102, 202 and presented to the call management computer 101. Whenthe calls arrive even if all arrive at the same time, their trunkinterfaces 203, 206 and the assigned DSPs 108 receive and answer ALL thecalls, determine the called party DID, ISDN or otherwise and determinethe call type, voice, Fax or data.

For Fax or data calls, the call management computer instructs theattached DSP 208 to establish a FAX or data call session and to receivethe transmission. When complete and stored on the Call ManagementDatabases 215, the call management computer 101, 201 alerts the calledparty to the new Fax or data files.

For each voice call received from outside, the call management computer101 with assigned DSP 208 proceeds with proactive caller identification,checks for applicable VIP rules and alerts the called party to the calleven if other calls are currently active or waiting.

Thus, for each system user only “One Number” is needed to receive allvoice, Fax and data calls from outside or inside with the CallManagement System able to identify and handle each type appropriately,even if multiple calls for the same party arrive concurrently.

This “One Number” ability of the Call Management System removes thetypical requirement for each user to have expensive, separate telephonelines and equipment for each different type of call and it also preventsthe conventional “busy” signal being received by callers, improvingefficiency and obviating starting telephone tag.

6. Proactive Caller Identification

Proactive Caller Identification is the means whereby the Call ManagementSystem augments and improves central office-delivered calling partyidentification. Even with no central office-delivered calling partyidentification, Proactive Caller Identification can identify the calledparty. Call Management System configurations are provided fororganizations which can include one or more forms of called partyidentification from their telephone providers and those which cannot.However, even for those which can, the correct caller identificationusing central office-delivered information occurs for only a modestfraction of all calls, not from blocked lines, pay phones, cellularphones, etc. thus, proactive caller identification is required in anycase.

If no or unusable calling party identification is provided by thetelephone service provider, A Proactive Caller Identification of theCall Management System requests the calling party to provideidentification of the caller to the called party. Call managementcomputer 101 utilizes a DSP 108 to access a call management database 215message to be provided to the calling party 118. A typical message is:

“So that I may inform Mr. Johnson of your call, please enter your homeof office telephone number or state your name.”

The calling party identifies himself through one of several means:

telephone keypad entry of an identifiable telephone number;

telephone keypad entry of an identifiable business number with attachedextension number;

telephone keypad entry of a unique assigned PIN number; or speakinghis/her name with subsequent voice recognition by the call managementcomputer.

Proactive Caller Identification is described using FIGS. 1 and 2.

A typical voice call to a system user might well originate from anoutside caller at a payphone 118. The PSTN 100 routes that call to thecentral office 103 which, in turn, presents the call to the CallManagement System via the CO trunks 102 and through the trunk interface203, circuit switches 204, and telephony signal buses 210 to the DSP208.

For businesses, the forms which CO-provided calling party identificationtake are limited, although growing. These include:

1. Caller ID containing the telephone number and/or name of the billedparty in FSK;

2. ISDN providing both the calling and the called party's numbers aspart of the “D” channel communications;

3. ANI DTMF or FSK provided by inter-exchange carriers with some 800/900services;

4. BCLID Bulk Calling Line Identification, in which a BCLID data linefrom the central office is used by the central office to provide thecalling number;

5. Transmitting the calling number along with the DID number in eitherDTMF or FSK form.

When a call is received 501, two different and parallel functions arestarted, identification of the called party 502 and identification ofthe call type 503. The calling party will be identified. Identificationoccurs through receipt of the calling party's extension DID after “wink”or T-1 in-band signaling or telephone number ISDN “D” channel signal orvia other signaling means.

When the auto attendant mode is used, the DSP 108 accesses the callmanagement databases 215 to play a message to the caller 118. Entry ismade of the called party's extension number or name encoded from thetelephone keypad or the name of the called party may be spoken and thenrecognized utilizing conventional auto attendant steps.

If the called party is not a system user 504, the PBX 104 is used toprocess the call in the form required by the PBX trunks 105 and the callis switched over switches 204 via the telephony signal bus 210 to thespecified number for conventional treatment

If the called party is a system user 505, the calling party's telephonenumber is determined 506 automatically through the receipt of a name ornumber from the list above.

If a caller identification is received 507 the identification number iscompared with the Primary Caller Identification Database.

If no automatic detection of caller identification occurs 508, thesystem provides a pre-recorded message to the caller 509 such as the oneabove. The requested information is received by the attached DSP 208 andused in subsequent identification.

If no response 511 is received the caller is considered “unknown” 521.

If the name was spoken 512, the call management computer 101 comparesthe name with entries in a voice identification database 214. If thename corresponds to one in the voice identification database 214, thecalling party is identified 521. If the corresponding name isidentified, the caller is “unknown”, the recorded name is retainedduring the call, in case the user wishes to have the name added to thevoice identification database 521 for future calls.

If a telephone keypad entry was made 513, the entry is used as an indexinto the Primary Caller ID Database 515.

If a match is found in the Primary Caller ID Database 516, the caller isidentified as the party in the matching database record.

If a match is not found in the Primary Caller ID Database 517, the entryof a telephone number is used as an index into the Secondary Caller IDDatabase 518.

If a match is not found in the Secondary Caller ID Database 519, logicgoes to step 521 with the caller considered “unknown”.

If a match is found in the Secondary Caller ID Database 520 the calleris identified as the party or business matching the Secondary Caller IDDatabase entry

Step 521 represents the end of proactive caller identification. At thatpoint, the call is handled according to any appropriate VIP rules 521and/or the called party is alerted to the presence of the call 522.

This basic procedure can be accomplished with many variations whichprovide the same results but may add, move or remove various steps toaccomplish it, e.g., obtaining information from the caller can be donethrough a series of different requests and responses, not just theefficient single one described above, or auto attendant identificationof the called party can be done following caller identification, insteadof before.

Additional Proactive Caller Identification capability is provided to acalled party once a call is received whether identified or unknown, by“double-clicking” a toolbar button requesting “More Identification”,e.g., a caller identified only as from “General Motors” may give toolittle information to the called party. Selecting the button to request“More Identification” causes Proactive Caller Identification to requesta different number from the caller so that he may be more closelyidentified as “Mr. Jones” calling from “General Motors”.

Voice calls arriving through Internet or other similar digital networksare identified using a “form” presented to the calling party, which hefills out providing the needed calling party information.

7. Continuously-Improving Caller Identification Databases

7.1 Calling Number Databases

Identification of a calling party, e.g., party 118, from the callingparty's telephone or PIN number or other entries is accomplished withtwo Caller ID Databases which are part of the overall Call ManagementDatabases 215. A Primary Caller ID Database is dynamic and continuouslyupdating. It includes names, telephone numbers and/or affiliations ofcallers to the organization, including employees, and is automaticallysearched by the call management computer 101 first as soon as a numberis known or through Proactive Caller Identification. If a match isfound, the name from that entry-is used by the Call Management System insubsequent VIP processing or to alert the called party.

The Primary Caller ID database contains specific entries relevant toindividual system users, e.g. for system user John Adams, theautomatically identified business number from one site of General Motorsis assigned to Mr. Jones, but for a different system user, Sam Archer,that same General Motors site number is assigned to Sarah Smith. Sinceboth entries will match the automatically identified number, the onematching the called party will be chosen. A further extension of thisprocess includes numbers not assigned to an individual system user, butto their group or even to the entire organization. All of these arematched to the calling party and the one most appropriate for the calledparty is chosen. One reason for this process is that all calls from anentire business site are customarily identified by the billing numberfor the site's PBX, not by individual. Thus, this procedure increasesthe probability of correct identification of callers based upon who theyare calling.

A second variation includes alerting the called party using all of thematching names and allowing the called party to select which oneactually is calling, removing the others.

If no match is found in the Primary Caller ID Database, the SecondaryCaller ID Database is searched. It contains a commercially availablelist of individual and business names and telephone numbers or anextraction from such a list. If a match is found to the Secondary CallerID Database, the name and/or affiliation and number is transferred tothe Primary Caller ID database to be used for subsequent calls. If nomatch is found, the caller is finally considered “unknown”.

With or without a call present, the called party, such as user 111 oruser 113, may use his workstation's call control window 115 to update orcorrect the Primary Caller ID Database for a caller or add a new callerto the database using any of the following indexes:

1. Home telephone number;

2. Business telephone number;

3. Business telephone number plus the person's extension;

4. Special PIN numbers assigned to or selected by callers;

5. Voice mailboxes for users;

6. Special coded entries including the * and # keys; and

7. Other indexes.

With or without a call present, the called party 111 or 113 may use theworkstation call control window 115 to update or correct the PrimaryCaller ID Database for a caller, or add a new caller to the database.

7.2 Voice Name Identification

Voice name identification is accomplished through a comparison ofpre-recorded spoken names in a Voice Name Database whichcross-references the person's name, affiliation and phone number similarto the Primary Caller ID Database.

The user can have a current caller speak their name, which is thenrecorded by the DSP 108 and stored by the call management computer 101,201 in the Voice Name Database for future use. Alternatively, if thecaller had spoken their name at the beginning of the call whenrequested, that saved recording may be used for the Voice Name Database.

Thus, the Primary Caller ID Database and the Voice Name Database arecontinuously updated both automatically and through user action,becoming ever more effective in identifying callers.

8. Call Notification & Control Via the Digital Network WorkstationComputer

The Call Management System provides intelligent call management throughwhich calls are handled by called parties using their workstationcomputer, not the telephone instrument as with conventional business PBXor other telephone systems. A called party, such as user 111 or user113, controls one or many concurrent calls directly through a callcontrol window 115 displayed on a workstation 114. Section 1 describesthe actions taken by the call management computer 101 and therelationships of calling parties and called parties for many of thespecific user control examples. This section describes the userinteractions at their workstations 114 using the many aspects of theirgraphical user interface call management window 115 and its subsidiaryscreens. FIGS. 6 a-6 e show the call management window and selectedsubsidiary screens. It is understood that different combinations andorganizations of screens, layouts, buttons, etc. can be configured toprovide the user his many Call Management System features and functions.Thus, the implementation shown and described is but one of manypotential graphical user interface layouts which could be used toimplement the user aspects of the Call Management System.

8.1 Call Notification of the Called Party

When a call arrives for a system user, the call management computer 101,in concert with its DSP processors 208, identifies the called party, thecall type and the calling party as described in Sections 1, 6 and 7. Forvoice calls, the call management computer 101 reviews any applicable VIPrules from the Call Management Databases 215 and, if none apply, todivert or affect the call, and the called party is in an appropriate“available” status, it notifies the called party 111, 113. Notificationmessages are sent through the digital network(s) 109 to the user'sworkstation 114. Notification of the user is accomplished through a callcontrol window 115 on the user's workstation 114 which “pops up” when acall arrives; a flashing icon which, when double-clicked, launches thepopup call control window; a special sound from the workstation alertingthe user to activate the call control window; any combination of theseor other alerting mechanisms.

\This notification is not the conventional telephone “ring” typicallyused by existing telephone systems. Instead, it uses the separate,independent and high-speed information path of the digital network(s)109 from the call management computer 101 to a called party. Inaddition, notification conveys significantly greater information to theuser and enables an entire array of new features and capabilities notpreviously available.

FIG. 6 a shows a basic call management window 115 user screen, as seenwhen it pops to the front on the user's computer display due to any ofthe alert reasons listed above, or because the user activated itdirectly. FIG. 6 b is a fully expanded user screen as might beconfigured by a “power” user providing more readily available functions,but in a more “busy” environment. The call management window FIGS. 6 aand 6 b supports a number of subsidiary windows through 6 c-6 e designedfor specific purposes and described elsewhere.

The computer program behind the call management window is kept active atall times when the user is in any of the “available” states. It isdesigned to be consistent and compatible with Microsoft Windows andother appropriate conventions for its overall look and feel, menuconventions, buttons, borders, help, etc. A user can activate differentfeatures in a variety of ways including the following, all of which arereferred to as “selecting”:

1. clicking the primary mouse button on the selected button or item;

2. clicking on the associated menu, such as “Calls” and then clicking onthe selected item;

3. double-clicking call notifications to transfer to that call;

4. clicking the secondary mouse button on a call notification to bringup a short menu of available options and then clicking on the selectedone; or

5. typing in the associated keystrokes such as “Alt-A” for the “answer”button.

The main call management window 115 screen 6 a is logically broken intothe following areas:

8.2 Customer Logo

On the top left of the screen is the customer's or vendor's logo 601 asshown. The logo can be changed at any time by providing a new graphicfile to replace the existing one.

8.3 User Status

The elongated button just below the logo is the user's status button 602which the user may select to change his status to the system. Userstatus includes:

1. “available to receive all calls”;

2. “available only for VIP calls”;

3. “unavailable—transfer calls to another location”;

4. “unavailable—until 4:00”.

5. “unavailable for all calls”.

For the fourth listed status, when the time occurs, the status wouldautomatically change back to “available”.

The last listed status provides a list of options such as “Out toLunch”; “In a meeting”; “On vacation”; “On a sales call”; “With acustomer”; and others. Alternatively the user may type in or modify oneof the standard options to provide more useful information for otherusers (see Section 17). Examples include “Out to lunch til 2:00”;“Giving a demo til 4:00”; “Out of the country til July 21, sendeverything to Judy”; and others.

8.4 The Message Board

Voice mail 606, along with Fax and data messages 603, e-mail 605 and“Flash” Notes 604 are “historical” messages, and are treated differentlythan “real-time” calls. For these historical messages, the user's callmanagement window contains a “message board” area just below the userstatus button with an appropriate control button for each type ofmessage 603-606. Whenever a message is available to be reviewed, e.g., anew Fax message, the Call Management System highlights the appropriatebutton name in flashing red and places a number adjacent to the buttonindicating the number of such new messages available to be reviewed.This procedure separates the various types of historical messages andsubstantially simplifies reviewing messages by users.

8.5 “FAX” Notification

When a user has received a new Fax transmission, the “FAX” button ishighlighted and the count of new Faxes is provided. Selecting the “FAX”button launches the FAX selection subscreen. That screen shows theuser's list of Faxes and summarizes the total number of Faxes, andidentifies the number which are new messages (see Section 13).

The user may select one or more of the Faxes to be viewed or handled inother ways. Also, the user may select a checkbox to limit the display toonly new Faxes. If the user selects a Fax to be viewed, the computeroperating system's FAX viewer is launched with the name of the selectedFax file, popping the selected Fax up in front of the other windows.

8.6 “Flash” Mail Notification

Flash Notes sometimes called “Flash” mail, are quick, simple messageswhich may be passed among system users. They are the equivalent of an“electronic shout across the office” intended as an improved, electronicversion of the classical “pink slip” notes on which messages used to bewritten then carried to the intended party and presented to him. Theuser is notified of his unread flash notes by a highlighted “Flash”Notes button and an associated count of such unread notes. Selecting the“Flash” button allows the user to review his Flash Notes as shown inFIG. 6 c. The Flash Notes screen indicates the originator 660 and thenote itself 661, and allows the user to close it 662, reply to it withanother Flash Note 663, or to return the call 664 by making a voice callto the originator. Flash notes can be sent to one or to many systemusers at a time, selecting multiple users for the note.

8.7 “E-Mail” Notification

When a user receives a new e-mail message, the Call Management System isnotified by e-mail services assuming the organization's digital networke-mail services provides this capability. The user is notifiedimmediately of his unread e-mail by a highlighted “e-mail” button withassociated count of unread e-mail messages. When the user selects hise-mail button, it launches the organization's e-mail client program,allowing the user to review and read his new e-mail messages. Onespecial feature the Call Management System provides is the ability torecord a voice message and attach that message to an e-mail message forlater retrieval by the recipient. This feature is based on thecapability of the organization's e-mail system.

8.8 “Voice-Mail” Notification

When a user receives a new voice-mail message, the Call ManagementSystem is notified by voice-mail services, assuming the organization'svoice-mail services provides this capability. The user is notifiedimmediately of his new voice mail messages by a highlighted “voice mail”button with associated count of new voice mail messages. If the CallManagement System includes an integrated voice mail subsystem, the voicemail messages may be retrieved in any order from the list provided. SeeSection 16 for a further description. Otherwise, the user accesses hisvoice mail messages serially via his telephone instrument, as isconventionally done.

8.9 User's Call Status

Near the top of the call management window 115, adjacent to thecustomer's logo, is provided the call status of the user 607, to whom heis currently connected, and the length of time the call has beenconnected 608 to the user. The length of time the call has beenconnected to the user is different than the length of time the call hasexisted as shown in the “time” field of the call alert box below. Whenthe user has selected an “unavailable” status, the message on the screenis that selected or created status message.

8.10 Call Alert Box

The call alert box is immediately below the user's call status, and itdisplays all real-time call alert messages for the user to control hisreal-time calls. The components shown on the screen for each callinclude: “CALL STATUS ICON” 611 sometimes flashing or moving whichindicates the status of each individual call; and a “CALL STATUSSTATEMENT” 612 showing the individual status of each call. The callstatus statement may state that the status is “ringing”, “connected”,“holding” or “recalling”. “Ringing” indicates the call has been receivedby the call management computer and processed, but has not yet beenanswered by the user. “Connected” indicates the call has been acceptedby the user, put through by the call management computer 101, and iscurrently the “active” call. “Holding” indicates a call has beenpreviously connected to the user and then placed on hold. “Recalling”indicates a call has been holding for longer than a predefined length oftime, typically 60-120 seconds, but the call is still “holding”.

Also displayed are the following: TIME 613 which is the time since thecall was answered by the call management computer 101; CALLER 614 toidentify the name and/or affiliation of the caller or “Unknown”; andNUMBER 615 to identify telephone number from which the call was made,when known, including: telephone numbers, “Internet” and otherapplicable identifiers.

“Source” identifies the source of the call as “outside”, “inside”,“Internet”, or “other”.

FOR “group” handling or employees who have temporarily moved to a“meeting” environment, described below, an additional field is displayedshowing the initials of the called party.

This call information display occurs even while other calls for the samecalled party are in process, allowing the called party to know who iscalling and to apply appropriate priority to each call.

When a call noted as “Unknown” is received, the called party may speakto the caller and determine who they are. The called party may choose toenter that person in the Primary Caller ID Database so that they areidentified properly the next time they call; or have them speak theirname for recording in the Voice Name Database for future calls; orsimply type the caller's name and/or affiliation as part of the callinformation for use as the call is transferred to others in theorganization.

In any case, the call information is corrected in view of the knownidentity of the caller.

8.11 Workstation Real-Time Call Controls and Management

A called party can control one call, or many concurrent calls, directlythrough a call control window 115 on a workstation 114. Section 1describes the actions taken by the call management computer 101 and therelationships of the caller and called parties for many of the specificcases. This section describes the user's interactions, features andfunctions for call alert, control and management. A user can exercisecall controls by any one or a combination of the following or otheractions:

1. clicking the special call-control buttons shown just below the callalert window 616-620;

2. typing the letter underlined for each button or menu item, e.g.,Alt-A for “answer”;

3. opening and selecting from the “Calls” menu using the mouse or typedcontrol letters Alt-C;

4. double-clicking the call to be answered see below;

5. secondary clicking the call to be handled to bring up a list of callcommands, then choosing the desired command;

6. dragging and dropping a call alert onto a call function button; and

7. using the cursor keys to move among call alerts.

User controls may be exercised via the specialized call control buttonsplaced just below the call alert box, or call controls may beselectively activated from screen menus, or from the special “ButtonBar” shown on FIG. 6 b. These controls and management functions include:

1. ANSWER 616 answers the selected call. If another call was active atthe time, it will be placed on “Hold” or “Hang Up” according to theuser's selected options.

2. HOLD 617 places the call on hold, disabling both inbound and outboundvoice paths, and starting the “recall” timer.

3. TRANSFER 618 transfers the caller to anywhere in the directorydescribed below, inside or outside the organization, or to a typed innumber anywhere in the organization or accessible via the PSTN (seeSection 11). For calls transferred from secretaries or telephonereceptionists providing “group” support, the default for transfer is theperson to whom the call was originally intended. Also, when a call isactive, the individual “Speed Dial” buttons or page of Speed Dialbuttons (see “call origination” below) become different “Speed Transfer”buttons for single-click, rapid transfers.

Transfers can be directed to any entry in the directory (person,location, inside, outside, etc.). Transferred calls are identified tothe transferred party with the initials of the transferring party asfurther information for call management. When transferring a call, thecall can be “Tagged” with a digital message or it can be transferredwith an appended voice message selected by the user from his drop-downlist and played out at the time the call gets “Answered” by thetransferee.

4. HANG UP 619 hangs up the call, releasing the caller and dismantlingthe caller's part of the call path.

5. VOICE MAIL 620 sends the caller to the user's voice mail box (seeSection 16).

6. DIRECTORY 630 opens the directory for update or use (see below).

7. CALL LOG 631 opens the call log screen and displays the call log ofthe user's calls (see Section 14).

8. VIP RULES 632 opens the VIP rules screen for update (see Section 10).

9. GROUP HOLD 633 sends the selected call, not necessarily the activecall, with any attached “Tag”, to all members of the selected “specialtylist” for servicing by the first available member, potentially includingthe user (see “specialty list” description below).

10. “FLASH” NOTE 634 creates a “Flash” note for sending to another user,as described above and in FIG. 6 d. Flash notes can be sent to one ormany system users at a time (selecting multiple users for the note).

11. TAG CALL 635 adds an electronic message to the selected call, notnecessarily the active call, which will display on the user's call alertbox and on any other system user's call alert box to whom the call istransferred or conferenced (see Section 12). An example of a “Tag” isshown in FIG. 6 b attached to the second call 645 in the call alert box.

12. “BOMB” SELECTED CALLER 636 creates a VIP rule, or applies to thecaller an existing VIP rule, which handles the current and all futurecalls from the selected caller, e.g., the “cold-calling broker”, in apredetermined manner.

13. “VIP” SELECTED CALLER 637 creates a VIP rule, or applies to thecaller an existing VIP rule, which elevates the selected caller to aspecific VIP status, by providing distinctive ringing, follow-merouting, etc. (see Section 10).

14. “PIM” ACTIVATION 638 links to a selected Personal InformationManager or database with the information about the selected call, notnecessarily the active call, in order to provide “Screen Pop” of theinformation associated with the caller. Note that if multiple PIMs ordatabases are used, a pull-down selection list is provided for the userto select the appropriate one to use for this specific call. Thisfeature can be done either by clicking the “PIM” button or when the callis answered for automatic “Screen Pop” (a user preference).

15. MUTE 639 disables the outbound voice path, retaining the inboundvoice path of the active call.

16. CONFERENCE CALLS conferences the selected call, not necessarily theactive call, with the selected party or group of parties from thedirectory or as typed in by the user. This includes anyone internally,working at home, or external to the organization anywhere in the PSTN.

17. RECORD CALL records the active call for later playback and analysis.

18. PLAYBACK RECORDED CALL plays back a recorded call selected from thelist of such calls in its own subscreen. Playback is done via a “VoicePathway” created to the user's telephone system 106 or the “Data Path”transferring the voice message to the user's workstation to be playedout via the workstation's voice capability.

19. MORE ID causes the call management computer 101 to play out a voicemessage from the Call Management Databases 215 asking the caller foradditional information with which to identify him, including: entering adifferent telephone number or stating his name. Once done, the call isreturned to the called party.

20. STATE CALLER'S NAME plays out the caller's name for the selectedcall, not necessarily the active call, as spoken by the caller duringproactive caller identification or More ID interaction phases. The callmanagement computer 101, 201 uses or creates a voice path to the calledparty and states the caller's name as spoken and recorded or uses the“Data Path” to transfer the voice message to the user's workstation tobe played out via the workstation's voice capability.

21. PLAY MESSAGE selects a pre-recorded message from the pull-down listof messages and possible actions to be played out to the caller, e.g.,“I am tied up at the moment and am transferring your to Sam”, and anaction to be taken after the message is played out, e.g., “Transfer callto Sam or return call to me, etc.”. In effect, this is the creation of atemporary VIP rule for use with the highlighted call.

22. VIP RULES selects a VIP rule from a drop-down list to be applied tothe highlighted call.

21. TRANSFER USER transfers the user from the current “available” stateon this computer to the computer associated with the selected otheruser, and changes the current computer to the “Unavailable” statusallowing the user to type in a reason for other users (see “meeting”environments described below).

22. USER PREFERENCES. The user may modify his screen and operationalpreferences at any time by selecting items from the “User” menuincluding a “Preferences” subscreen, allowing the user to change any ofa series of his own operating characteristics including displaying the“Button Bar” 630-639 of FIG. 6 b; displaying the user's “Speed Dial”buttons 640; displaying the call management window's “Status Bar” 641showing the user's name, the current date and time; and whendouble-clicking a call alert, place the original active call, if any, on“Hold” or “Hangup”.

User control messages resulting from user interactions with the callmanagement window 115 and its subsidiary screens are returned, via thedigital network(s) 109, to the call management computer 101 where theyresult in the requested actions being performed, and/or appropriateinteractions with the Call Management Databases 215.

8.12 “Directory” Support

One important aspect of the Call Management System is support for the“Directory”, both corporate and personal. The directory user accessshown in FIG. 6 d contains:

1. ENTRY 681. The name/affiliation of a person or organization includingemployees, voice mail, places, e.g., “conference room”, outsider.

2. NUMBER/STATUS 682. The received telephone number or status of theentrant, including local numbers without area codes; long distancenumbers with area codes; international numbers with country codes, citycodes, etc.; Internet access; status for system users; and other.

3. RETURN TELEPHONE NUMBER The telephone number to call that person, ifdifferent from above.

4. VIP STATUS defines this person as having VIP status.

The user maintains the directory through subscreens to add “New” 683entries, to “Delete” 684 an existing entry or to “Edit” 685 a selectedentry. The user may define an entry as “Corporate”, available to allusers, “Group”, available to users specified as part of a specified“group”, and “Private”, available only to the user himself.

When the directory is opened, the user may select how it is sorted,limited, and displayed, including internal entries only, externalentries only, private entries only, sorted by name, sorted byaffiliation, sorted by number, limited to area codes, or ranges ofnames, and others.

Access to entries is conveniently available by scrolling the sidebar 686using the mouse, scrolling down or up the entries using the cursor keys,clicking letter buttons 687 to jump to the start of names beginning withthat combination of letters—after a limited time, three seconds or so,the list is cleared and the user may start again. As the user types thename, each letter entered causes the list to jump forward to the firstof the names beginning with the list of characters entered. Once a matchhas been made, the user can key “Enter” or click “Call” to originate acall to that number, or the user configuration can be set so that thenumber is automatically dialed as soon as a match is made. After alimited time, three seconds or so, the list is cleared and the user maystart again; and others.

The directory is used throughout the call management window 115functions for routing calls VIP rules, originating calls 688, reviewinguser status, transferring and conferencing calls, sending “Flash” notes,and many other things.

While the directory is a part of the Call Management Databases 215, itis used in a wide variety of ways and with entry sorting and display asappropriate to each.

8.13 Call Origination

The Call Management System provides flexibility and user convenience for“originating” calls, in addition to its features and functions forhandling “incoming” calls. The user may originate a call at any time,even while other calls are active or waiting, by opening the “Directory”described above, selecting an entry and clicking “Call” or pressing“Enter” as described above; opening the directory described above,typing a name, and having the system automatically begin dialing when aunique match has been found or waiting until the user clicks the “Call”button as selected for each user; clicking “SpeedDial” buttons 640defined uniquely for each individual from the “directory”. Multiplepages of SpeedDial buttons are provided for users requiring rapid accessto a large number of dialable numbers; or activating the “Dialpad”button 642 to bring up the dialpad subscreen; utilize the SecondaryCaller ID database to find a number and/or name, then outdial it.

The “Dialpad” subscreen, shown in FIG. 6 e, includes a representation ofthe telephone dialpad with the numbers 0-9, # and * 690. Calls may beoriginated by any of the following:

1. Clicking on the dialpad buttons to enter a dialable number thenclicking on the “Dial” button. Note that the “flip” button allows theuser to “flip” the dialpad so that it matches an adding machine keypadrather than a telephone keypad. Once entered, numbers may be removed oneby one by the “Back” button 691, or cleared altogether 692. “Pause” 693codes may be entered for use with modems, dialers, etc.

2. Typing in a number from the keyboard and then the “Enter” keyboardkey.

3. Clicking the “Redial” button 694 then selecting any one of the lastfive called numbers shown.

4. Activating the “Directory” 695 and choosing an entry to be called.

5. Typing in a name from the directory and having the system dial thatperson when a match is made.

When outdialing a number, the Call Management System provides theability to “Redial” repeatedly until the call gets through an otherwisebusy line.

When no call alerts are present and no user calls pending, the callmanagement window 115 shows a “Recall” button and a “Directory” buttonin place of the specialized call control buttons below the call alertbox. Clicking them brings up the “recall list” or the “Directory list”as described above, simplifying call origination.

8.14 “Outside” Employee Support

The Call Management System supports “Outside” employees who are rarelyor never in the office and who have no workstation. Such “Outside”employees are given “pseudo” numbers for which no actual workstation ortelephone extension exists, which may be used by callers. The “Pseudo”user establishes his own VIP rules which automatically handle all hiscalls, including his own voice messages, for selected callers andre-routing of his calls to other users.

Thus, the “Outside” person is supported, just like other system users,with callers being unaware that person is not in the same location.

8.15 “Group Secretary” Support for Calls to Specified Groups ofEmployees

The destination party for calls does not have to be the actual calledparty. The Call Management System provides for a specific user to handlecalls for a select “group” of others, system users or not. This “group”handling mode is intended for a secretary or telephone receptionisthandling a group of executives or others, and for an organization'stelephone operator. In either case, the specified user is expected toreceive calls, prioritize them appropriately, respond to eachaccordingly, attach “call tags” and transfer calls as needed, thuseffectively screening calls for the specified group.

The basic difference between direct notification of a system user andnotification of the “group secretary” receiving calls for theindividual, is that the call information displayed on the callmanagement window is expanded to include the initials or otheridentification of the actual called party, allowing the secretary toknow to whom the call was destined, and handle each call as appropriate.This capability requires appropriate information in the call managementdatabases, identifying the group of people, the secretary to answer thecalls, the initials or other identification for the called parties andother control information. On a dynamic basis, system users can be addedto or removed from “Group Secretary” mode by those assigned “Secretary”status, allowing for the cases where a user runs by the secretary on hisway out saying “I have to go, please handle my calls”.

8.16 “Meeting” Support for Users Away from their Workstation System

Users typically use their own workstation computer for most of theiractivities, but occasionally users need to leave their workstation andmove to another location for a meeting or otherwise. Users can move toanother location and, using the call management window on the computerwhere they are currently active and logged in, they may log off andchoose an appropriate “unavailable” state, or select the “TRANSFER USER”function and move to the new site. Once there, they may log on to thecall control window of the computer at that site, along with others whomay be sharing the meeting site, or the person whose computer is at thatsite. When this happens, a temporary “group” definition is created andthe users who are at that site sharing the site's computer, are treatedas described above, with their call notifications sent to the site'scomputer as though a secretary or operator were screening the calls eventhough the individuals may be managing their calls themselves.

This procedure allows users to hold meetings in offices or conferencerooms while continuing to receive and handle their calls as appropriate.

Users who attend meetings outside the office may create the samecapabilities by using their portable or other computer at the outsidemeeting site, calling into the Call Management System via Internet orotherwise, and providing the telephone number for the meeting site. Withthis, one or more users can continue to receive and handle callers asthough they were in the office.

8.17 “Specialty-List” Support for Special User Groups

The Call Management System provides the ability to define multiple“Specialty lists” of users who work together or have certain affinities,e.g., sales, accounting, customer service, manufacturing, etc. Thesegroups represent arrays of users who can receive and handle certainkinds of calls, such as sales calls, customer service calls, etc. Eachsuch special group is assigned a pseudo extension number through whichit may be reached. The purpose of these “specialty lists” is to providea new means for rapidly getting calls to the first available member ofthe group able to take the call.

When a call is made directly to the pseudo number or, a call istransferred to the number, the call management computer 101 concurrentlyalerts all of the users in the group by sending alert messages down thedigital network(s) 109 to their workstations 114 and thence to theircall management windows 115. The first person in the group to “answer”the call causes the call management computer to connect the call to theanswering party and to remove the call notification from the callmanagement windows of the other members of the group.

Specialty List Handling is a means to provide rapid access to thedesired function within an organization, without having to go through anoperator as is otherwise customary.

8.18 Feature Activation

These many features can be turned on individually for each user,maximizing the effectiveness of the organization's training and callmanagement.

8.19 TAPI Client

The workstation software also provides a TAPI client for otherapplications which need to outdial or receive calls, e.g., this featuremeans a PIM can place a TAPI call to have a number automatically dialed.

8.20 Automatic Updating

The Call Management System automatically updates the user's workstationsoftware from the call management computer whenever changes are needed.This is invisible to the user and is handled automatically at user logontime.

9. Multiple Call Handling Using a Single Extension

The Call Management System allows each user to be aware of and to managemultiple calls at the same time, using a single extension number, andusing a simple POTS telephone instead of the customary expensive,multi-button, proprietary telephone instruments 106. By using thedigital network(s) 109, workstation computer 114 and call control window115 for communications with the called party 111, 113, the CallManagement System is freed from the conventional limits on when, how andhow much information can be communicated to the called party andlikewise, when how and how much control can be exercised by the calledparty.

The Call Management System presents multiple calls, as they arrive andare identified, (Sections 6 and 7) to the called party 111, 113 usingthe call management window 115. Multiple calls are identified to theuser through call alert information which is displayed in the call alertbox of the call management window (see Section 8). The called party,applying his own preference, may then deal with each call as mostappropriate. Calls can be: answered, placed on hold, sent to voice mail,transferred to an individual or “group list”, recorded, hung up or anyof many other appropriate actions. All of these capabilities are underthe control of the called party, and all can be exercised for any of thecalls, thereby giving the user complete freedom to manage their calls asbest fits their ever-changing priorities, e.g. switching from acold-calling broker to an important customer.

The called party can effectively interact with multiple callersconcurrently by rapidly “switching back-and-forth” as needed amongcalls. The called party simply double-clicks a call, other than the onethat is currently active, and the call control window 115 causes amessage to be sent down the high-speed digital network(s) 109 to thecall management computer 101 which instructs the circuit switches 204 totransfer the voice path of the new call to the re-usable voice pathway121, 221 to the called party and leave the previous connected call onhold or hang it up as defined by the called party for their own specificconfiguration. In this way, the called party can “bounce back-and-forth”rapidly among many calls, servicing them as priority dictates. Thiscapability is in sharp contrast to the time-consuming creation andtearing down of entire connections each time, as required byconventional PBX and other switch systems 104.

Thus, handling multiple calls at the same is simply, logically andrapidly controlled by the called party, using the computer's mouse orkeyboard while repeatedly reusing the existing voice path.

10. User-Defined VIP Call Handling

VIP call handling is an automated adjunct to or an alternative for thecall control window 115 notification of the called party 111, 113 andcorresponding workstation control commands. With VIP call handling,specific callers, groups of callers or all callers are given uniquetreatment based on “VIP rules” defined by the organization and/orindividually by the user. Call management window 115 “VIP Rule”subscreens allow the user to manage their own VIP rules. The VIP rulesubscreens of the current implementation are shown in FIGS. 7 a-7 c. Aswith the other aspects of the user interface, this is but one potentialconfiguration of buttons, selections and controls which can beconfigured to allow the user to create and manage his VIP rules. VIPcall handling rules for each user are maintained by the call managementcomputer 101, 201 as part of the call management databases 215 and aredynamically changeable by and for each user, except for corporate-widerules. Changes to the rules can be made through the user's VIP Rulesubscreens of the call management window 115 on their workstation 114,through the user's laptop or other remote computer attaching through adigital network or through the user calling and changing characteristicsthrough direct telephone entry.

VIP handling for special callers, customers, etc., is a majorimprovement in the efficiency and effectiveness of call handling fororganizations. VIP rules allow each user to tailor how callers arehandled and routed, and are especially effective for “follow-me” routingin which an important caller, or group of callers, can be assigned to arule which specifies that: when you are out of the office to callautomatically to your cellular, car or other phone. Later, the user cancall and change the rule to route calls to a different phone number.Additional features include, “find-me” capability where the user canhave several numbers automatically dialed, such as car, cellular, home,etc. to find the called party for direct connection to the caller.

The basic VIP Rule subscreens consist of a list of existing VIP ruleswhich may be selected 701; the ability to activate an additionalsubscreen to “Add” 702 another rule; the ability to “Delete” 703 theselected rule; the ability to “Copy” 704 a rule for later change; andthe ability to make each rule currently “Active” or “Not Active” 705.

Each VIP Rule contains three parts to define who the rule applies to,what action is to be taken, and when the rule applies. FIG. 7illustrates the three displays for the who, what, when parts of the VIPrules.

Screen 7700 is used to define “WHO” does the rule apply to. The rulewill apply to the caller that is highlighted on the display. The ruleapplies to specific individual callers 710 whether included in the“Directory” or just typed in; selected callers 711 from the directoryand/or typed in to whom the rule will be applied; or everyone who calls712, typically for use when the caller is “unavailable”.

Individual callers or groups of callers can be included in multiple VIPRules, allowing the user to specify different handling under differingconditions.

The Call Management System provides for multiple greetings is messagesto be played out to defined groups of callers based upon their callingnumbers, e.g., all callers from New York area codes get one message andthen get transferred to Sam. This provides special treatment for callersfrom different locations. Screen 7701 is used to define “WHAT” actionsare to be taken. VIP rules can result in one or a series of actions tobe taken:

1. Play out selected pre-recorded messages 715 to the caller,personalized for the caller and recorded in the called party's ownvoice, e.g., “John, I'm out of the office today. If you need to speakwith sales, press one, or to speak to Sam, press two. Otherwise, I willcall you back tomorrow.” Or “John, I'm on the phone right now, but don'thang up, I'll be right with you.”. These messages can also be used for“Callback” responses to important persons, e.g., “Mark, I got your FAXand have found the shipping date is December 12. Let me know if I canhelp you further.”

Creating new messages is done by selecting “New” from the list ofprerecorded messages, whereupon, the call management computer 101, 201will establish a voice pathway 121, 221 to the user's telephoneinstrument 106 and the user will record the message. When completed, theuser will name the message, edit it as needed and then have it availablefor the VIP rule being created or modified.

Special “Menu” messages can be created, e.g. “To speak with Tom press 1,to speak with Sam, press 2, to speak to my secretary press 3 or to speakto the operator press 0”.

2. Receive information entered by the caller via the telephone keypad,spoken or by other means.

3. Process the entered information, re-routing the call to one of aseries of other numbers based upon the entered information, e.g., “2” toreach Sam.

4. Use the entered information as a “Tag” to the call (see Section 12).

5. Activate another VIP rule based on information input by the caller.

6. “Forward” 716 the call to another destination anywhere inside oroutside the organization as typed in or selected from the directory.

7. Transfer the caller to voice mail 717 whenever he calls.

8. “Return to me” when the transferred call is complete.

9. “Follow-me” routing for certain callers allows the user to specifyone or a series of numbers to be used to reach the user when thespecified caller calls. The user switches from one number to the next bycalling his own extension and entering telephone keypad numbers tochange from one number to another, e.g. car phone, home phone, cellular.

10. “Find-me”, “Chase me” routing for certain callers allows the user toinstruct the Call Management System to call a series of differentnumbers simultaneously, attempting to locate the user. When foundthrough voice entry or information entered via the telephone keypad orotherwise, the caller is switched automatically to that number, e.g.,home, car phone, cellular, etc.

11. “Page me” or “Beep me” instructs the system to place a call to thespecified beeper service and page the user. It also includestransmitting the caller's name and telephone number for display on theuser's beeper, where that service is available. This is useful for bothreal-time calls which are not completed because the user is“unavailable” and for when calls are sent to voice mail. One variationof the “Page me” approach allows the caller to stay on the line whilethe page is made and then, when the caller calls his own “One Number”and identifies himself, to connect the two calls together.

12. Provide “Priority Ringing” 718, a special sound when this callercalls to alert the user to a VIP caller.

13. “Hang Up” 719 when this caller calls.

14. Place the caller on “Hold” 720 whenever he calls.

Screen 7702 is used to select “WHEN” the rule applies. The followingselections may be made for “When”:

1. For certain status of the called party:

always 730

unavailable for calls or not answering 731

available but on the phone 732

available for VIP calls only, etc.

2. Within defined days of the week 733.

3. Within specified times of the day.

Certain VIP rules are pre-defined and merely have callers added to them.These include:

1. “Bomb” the caller and send all his calls to user's voice mail.

2. “VIP” the caller, giving him special ringing and notices.

3. “Follow Me” having the system follow the user.

4. “Find Me” having the system find the user.

5. “Page Me” having the system page the user.

6. Others.

10.1 Temporary VIP Rule Usage

The user may highlight a call in his call alert box and then select arule from a drop-down list for immediate application to that call, eventhough the caller does not normally have that rule applied to him. Theuser may also create a temporary rule (play out a message and specify afollowing action) for handling a highlighted call.

The Call Management System further enhances the user's ability to handlehis important callers by providing the ability for the user, when out ofthe office, to call to the system and record new voice messages and/orchange selections for his VIP rules via entries from his telephonekeypad voice recognition or otherwise.

10.2 Advanced Message Notification.

The user may establish a set of rules that specify how the user is to benotified of various types of messages. For example, the user could setup a rule like, “If I receive an e-mail message from Victor between 5 AMand 7 AM, call me at my car phone; if I don't answer, page me.” Anotherexample might be, “If I receive a Fax from anyone at Motorola, forward acopy to my home Fax machine.” One of the Call Management System'sprimary capabilities is to function as a central repository for messagenotifications, where the user has complete control over the notificationmechanisms.

11. Routing Calls Inside or Outside to the Organization

Unlike existing PBX or other telephone switching systems, the CallManagement System routes calls internally within the organization orexternally anywhere in the PSTN. Thus, calls can be transferred by acalled party, or via VIP rules, anywhere the telephone network reaches.This is a major departure from the conventional telephone switchcapability, which is customarily limited to intra-organization routingonly.

When a call is received by the call management computer 101, the callingparty, called party, and call type are used to determine the neededaction. For voice calls, the called party may be alerted, receive thecall and transfer it to a specified number or VIP rules may specifytransfer to another number. It matters not that the number specified iswithin the organization or external to it. In either case, the callmanagement computer transfers the call as specified.

For internal calls, the call management computer establishes a voicepathway 121 to a destination extension, e.g., user 111, and alerts thecalled party if that destination applies to a system user. Fordestinations outside the organization, an available outgoing CO trunkcircuit 102 is seized or a two-way circuit is negotiated through which avoice pathway is established and the call path transferred.

Calls utilizing Internet voice capabilities are placed through theVoice-over-Internet interface.

The ability to route calls anywhere, not just within the organization,is a major improvement in efficiency for the organization.

12. “Call, Tags”

The Call Management System provides a mechanism whereby system users can“tag” calls with digital messages, which then remain with the call, butcan be modified so long as the call exists, no matter to which systemuser it may be transferred. Call tags are a unique ability, availableonly because of call management via the user's workstation computer forwhich digital information is conventional, unlike telephone instrumentswith their lack of or minimal display capabilities. These digital calltags provide an advanced and convenient means for one user to provideuseful information to other users, within the organization, about theneeds or interests of the caller, or anything else that may beappropriate, e.g., “Sam needs to know about system installation” or“John has a question re: pricing”.

A system user may “tag” any call showing in the call alert box of hiscall management window 115 with or without his answering the call, thuscreating a digital message which, along with the initials of the taggingparty, immediately appears attached to the call alert line. As that callis transferred to or conferenced with one or more other system users,the notification and attached call tag moves with it, always displayingalong with the call alert line. Any receiving system user can modify,expand or even erase the call tag message, as appropriate.

“Special Tags” can also be added to a call which provide for return ofthe call when completed by the transferee; return of the call if nottaken by the transferee; transfer of the call to another destinationwhen done; or others.

Call tagging is an electronic tool similar to the old-fashioned “pinkslip” notes of paper which were used to pass messages within anorganization. However, call tagging is a significant improvement, sinceit is digital, automatic and, specifically, ties the call tag message tothe call itself. Call tagging is a powerful new means to improveefficiency and information flow within an organization.

13. Facsimile Fax and Data Calls

The Call Management System converts every system user's telephoneextension into a never busy FAX gateway, with automatic detection andreception of incoming Fax messages, and confidential delivery to therecipient's desktop computer. No Fax hardware need be added at thedesktop, and no special Fax telephone lines or numbers are required toreceive Faxes, Faxes or data transmissions are simply sent to the user's“One Number” unique personal point-of-contact extension. The systemdetects and automatically receives each Fax or data transmission andelectronically delivered it to the user's desktop machine, without hisextension ever ringing or interfering with normal voice traffic(private, paperless, immediate delivery of electronic transmissions.

Notification of received Faxes or data files is via the desktopcomputer, similar to e-mail, and is also given by the system (in voice)when the user calls in to retrieve his voice mail. Outbound Faxes may becomposed in the desktop machine and passed to the system for private,paperless, immediate transmission to the recipient(s).

These features improve the efficiency, confidentiality, security andpunctuality of Fax and data delivery and transmission within theorganization.

13.1 Receiving Fax and Data Transmissions

During all calls, assigned DSPs continue to search for Fax or datasignals which, if encountered, immediately switch control to theappropriate Fax or data mode protocols. Thus, even during a voice call,the caller may switch to Fax “Send” mode, causing a CNG or otherappropriate signal to be sent. The assigned DSP 208 receives the signal,alerts the call management computer 101 and the assigned DSP is thenswitched to a Fax protocol mode to complete the transmission.

The call management computer 101 identifies an incoming call as a Fax ordata call type because the system received a CNG or other signal for Faxor an appropriate DTMF or other signal for data or a “reverse” modemcourier signal. If the fax or data call was dialed directly to a systemuser's extension number or the caller used the call attendant feature toidentify the called party, the call management computer 101 instructsthe attached DSP processor 108 to receive the fax transmissionsautomatically using a standard Fax protocol with adjunct filetransmission capabilities. For data transmissions, the DSP is instructedto use an appropriate data protocol Fax plus data, X.34, ISDN, TCP/IP orother. Even during a voice call, the caller may switch to Fax mode,causing a CNG or other appropriate signal to be sent, identified by theassigned DSP 208 and the DSPs to be switched by the call managementcomputer 101, 201 to Fax protocol mode to receive the transmission.

Because the Call Management System knows who is receiving each Fax, itapplies a unique Fax identification for each user uniquely identifyingthe receiver to the Fax sender.

When a transmission is received, Fax, file, video, etc., it is stored onthe call management computer's database 215 or on the organization'sdigital network(s) message storage facility. The destination party 111,113 is then notified of the new Fax or data through messages sent by thecall management computer 101 down the digital network(s) 109 to theuser's workstation 114 and finally to the user's call management window115 (as described in Section 8).

When notified, the destination party 111, 113 may review the list ofunread Fax or data messages and then may request that the Fax or datamessage be transported to their workstation 114 via the digitalnetwork(s) 109, from whence it can be viewed, printed, archived andtreated as any other such file (see below).

The result is that each system user is provided with private, paperless,and immediate Fax and data send and receive capabilities without havingto add any hardware to their desktop computer 114.

This use of a “One Number” direct user access with automated receptionof Fax and data messages is a significant improvement over otherapproaches, which typically require: a dedicated Fax/data line andhardware for each person's desktop computer 114 doubling the number ofCO circuits in use, or having all Fax and data messages routed to adesignated person for later sorting out and distribution, therebydelaying delivery and losing privacy, or assigning a separate block oftelephone numbers attached to a special fax/data transmission server.

13.2 “FAX” Notification

When the user has received a new Fax transmission, the call managementwindow's 115 “FAX” button is highlighted and the count of new Faxes isprovided. Selecting the “FAX” button launches the FAX selectionsubscreen FIG. 8 of the current implementation. That screen shows theuser's list of Faxes and summarizes the total number of Faxes and thenumber which are new 800. The list includes:

1. “Check” 801 if the Fax has been previously seen;

2. The date and time received 802;

3. From whom the Fax originated 803;

4. Subject of the Fax 804 for future reference;

5. “Print” the Fax document;

6. “Send” copy or original Fax documents to someone else from thedirectory or typed in with attached “Annotation” voice or digitalmessage and name of the original recipient;

7. “Broadcast” Fax document to selected people/locations form thedirectory or typed in with attached “Annotation” voice or digitalmessage and name of original recipient. Broadcast of Fax documentsincludes selection of dates, times, retries, etc.;

8. “Routing” one or more Fax documents to the system users assigned tothe routing list with attached “Annotation” voice or digital message andname of recipient.

The user may select one or more of the Faxes to be viewed 806, deleted805 or forwarded 807. Also, the user may select a checkbox to limit thedisplay to only new Faxes 809. If the user selects a Fax to be viewed,the computer operating system's FAX viewer is launched with the name ofthe Fax file, popping up the selected Fax in front of the other windows.Voice or digital “Annotation” by the system user is provided in the sameway that voice messages are recorded for VIP rules.

13.3 Unique Call Routing for Faxes or Data

Fax or data transmissions, received for specified numbers such as theorganization's base number, may be accepted for a specified user, e.g.,the organization's operator who sorts them out and sends them to theappropriate users, prints them and deals with them conventionally, orsends them directly to a Fax machine or computer for data prefacing.

13.4 Special Data Calls

For special kinds of data calls, e.g., video conferencing, the CallManagement System transfers the call automatically to an appropriateextension for handling by a specialized device.

13.5 Laptop Data Calls

The Call Management System provides for users to call into their “OneNumber” with their laptop computer and, after being identified to haveaccess to their desktop computers for file transfers and maintenance aswell as to have access to their e-mail and other electronic messages.The user also has access to their Call Management System functions (VIPrules, status, etc.) in order to change and update them as appropriate.

13.6 Outgoing Fax and Data Transmissions

Fax and data transmission of documents/files created at the desktopcomputer 114 is also provided through conventional “printing” to thefax/data feature of the call management computer, from whence thedocument can be transmitted automatically immediately, or at specifiedtimes, to one or more recipients in the Call Management System directoryor to numbers typed in by the user and with specified retry attempts.

Because the Call Management System knows who is sending each Fax, itapplies a unique Fax banner for each user specifically identifying thesender to the Fax receiver.

13.7 Retrieving Fax or Data Files Via “One-Call” Message Retrieval

The Call Management System also addresses the special needs of thetraveling or at-home user. Using any touch-tone telephone, the userretrieves his voice mail messages from the enterprise's existing voicemail system and, in the same call, is notified by system of theexistence of his unread Fax, data and e-mail messages. He can theninstruct the system to Fax these messages to a convenient Fax machinenear him, e.g., a hotel, airport or home Fax, or to transmit them to hishome or laptop computer (see Section 15).

14. User-Accessible Call Logs

All calls received by or sent from the Call Management System aresummarized in a user-accessible call log as part of the overall callmanagement database. This allows users dynamic access on demand to thecall logs, and enables the ability to return missed calls with only theclick of a mouse. The use of a user-accessible call log improves theability for system users to be aware of who called, to get back tocallers missed and to monitor their telephone usage. Management of theorganization may also use the call log database to monitorresponsiveness to returning calls and misuse of business telephone

All calls received or placed by the Call Management System aresummarized in the Call Log portion of the call management database. Eachuser has direct access to his own call log containing all his calls,even if the caller chooses not to leave a voice mail message. Using thiscall log, the user may simply double-click to return missed calls, withthe system automatically outdialing them.

Each user is provided a complete log of his own voice calls, availableon demand, through subscreens of his call management window 115, asshown in FIG. 9 a and with log sorting options in 9 b. As with all otheraspects of the user's graphical interface, different screens, buttons,wordings, etc. can be used as the means to accomplish the same ends. Thecurrent implementation as shown is but one of many possiblearrangements.

Each log entry includes the following fields:

1. CALL INDICATOR 905

“Blank” for incoming calls

“Red Dot” for missed calls sent to voice mail without answering,transferred without answering, etc.

“Out” for outdialed calls

“VM” for calls sent to voice mail

Others.

2. DATE 906 Date the call occurred

3. TIME 907 Time the call began

4. LENGTH 908 Duration of the call HH:MM

5. NUMBER 909 Telephone number from which the call originated

6. CALLER 910 Name and/or affiliation of the caller or called party.

7. “TAG” Associated “Tag” message and initials of the “Tagger”.

8. TRANSFERRED TO Number or name of transferred party for transferredcalls.

9. CALLBACKS Count of times a callback was attempted and the date/timewhen the callback was successful.

This directory is retained as a user-accessible part of the CallManagement Databases 215 and can be accessed at any time through theuser's call management window 115. When opened, the user can click the“Options” button 911 to bring up the “Options” subscreen 912 shown inFIG. 9 b. The user can define how his log is to be sorted by picking andchoosing among the various options as shown, including only unansweredmissed calls; incoming calls, outgoing calls, only new calls since theuser last was available, calls only within the last number of days,weeks, months, etc., and others.

If a call is active at the time the call log is accessed, the calls aresorted and limited to those to/from the caller of the active call,allowing the user to know when calls occurred between himself and thecaller.

Whenever the call log is accessed, any call shown can be“double-clicked” by the user to tell the system to outdial the callerimmediately. This allows users to keep track of their missed calls andto return calls quickly, e.g. after returning from lunch, even if thecaller failed to leave a voice mail message.

Even though each user is limited to viewing only his own log entries,the system manager may have the logs of all employees sorted, combined,processed and printed in any way management requires using theindustry-standard database tools as an added means for managing thebusiness. For example, it may desirable to know how long on the averageit takes for a customer to get a call back, or whether certain employeesare misusing corporate telephone resources for personal purposes.

Additionally, these call logs provide the means for corporate managementto do accounting comparisons with central office bills, verifyingaccuracy and completeness and often saving telephone costs. The logs canalso be sorted and compiled to show traffic flow for workplacemaximization.

15. “One-Call” Message Retrieval

The Call Management System permits for traveling or out-of-the-officeemployees to make a single telephone call to receive all of theirelectronic messages, whether voice mail, Fax messages, data files ore-mail. This feature is a significant improvement over the usualrequirement to make several calls, using different technologies toretrieve messages in different forms. Retrieving all of the user'selectronic messages during a single telephone call is a majorimprovement, especially from remote locations where telephone callconnections are difficult, unreliable or can take a long time toestablish, and re-establish.

One-Call Message retrieval is accomplished by the traveling system userplacing a call to the organization's voice mail 116, as is doneconventionally by calling the voice mail retrieval number. The callmanagement computer 101 recognizes the destination number as that ofvoice mail retrieval and puts the call directly through but remainsonline with the attached DSP 208 assigned to identify the caller'smailbox number, as entered through the telephone keypad or by voice.With the voice mailbox number, the call management computer 101identifies the caller 118 through correlating voice mailbox numbers inthe call management database 215.

The Call Management System then identifies the caller as a system userand checks to determine if any new Fax or data messages are stored inthe call management database, and determines if any new e-mail messagesfor the caller exist, assuming the e-mail system can report suchinformation. If any of these electronic messages exist, the CallManagement System plays an unobtrusive chime for the caller to hearsaying, in effect, “you have electronic messages in other forms waitingfor you, don't just hang up.” The user then knows to logoff from thevoice mail system 116 when he is finished.

When the assigned DSP 208 detects the user's entry of a voice mail'slogoff sequence, it informs the call management computer 101 which theninstructs the DSP to play out a message from the Call ManagementDatabase 215, e.g. “You have two new Faxes and three new e-mailmessages, press one for immediate delivery.” If the caller responds asrequested, the caller may then be asked for and respond with anadditional security code or his spoken voice which is verified.

The caller is then given a menu of delivery options for the electronicmessages, including:

1. Dial out and send them to my home computer

2. Dial out and send them to my home Fax machine

3. Dial out and send them to a convenient Fax machine at this number

4. I will attach my laptop computer and then send them to meelectronically now

5. Read part or all of the messages to me (via text-to-speech) before Idecide which to send

Depending upon the option selected by the user, the call managementcomputer 101 will respond accordingly including providing translationsas needed, e.g., e-mail sent to a Fax machine is converted to Faxformat, etc.

The system user may also call his own “One Number”, override thegreeting message and identify himself using his voice mailbox number andappropriate password. Once identified, the user is provided the sameretrieval capabilities as though he had called voice mail, withouthaving to go through the voice mail process described above

The Call Management System thus provides the means for a system user toretrieve all his electronic messages during a single telephone call tothe organization.

16. Voice Mail Handling

The Call Management System utilizes voice mail in four broadly differentways: transferring callers to voice mail, alerting system users to thepresence of voice mail messages, as a part of “One-call” messageretrieval (see Section 15) and as an integrated subsystem of the overallCall Management System, thereby becoming the organization's voice mail,providing expanded voice mail capabilities to system users.

16.1 Transferring Callers to Voice Mail

The Call Management System eliminates “voice-mail-jail”, to whichcallers are so commonly subjected, because only system users send callsto voice mail, not automated machines as in the past. Callers to systemusers are transferred to voice mail only because the user makes thatchoice directly or because of predetermined VIP rules

When a call is transferred to voice mail (see Sections 1 and 8), thecall is transferred to the organization's internal voice mail extension,kept in the call management database 215, and the extension number ofthe called party, also from the call management database, is entered bythe attached DSP 208 to tell the voice mail system, whether integratedwith the PBX or not, which voice mailbox to use. The call managementcomputer 101 records the length of time the caller uses the voice mailas part of the voice mail log, from which the called party can obtainsome additional knowledge about the caller's message, or lack thereof.Even if no message is left, the call log reflects to the user that thecall was received, allowing the called party to return the call usingonly the click of his mouse.

The Call Management System also provides a “fake” voice mail capabilityto which an annoying caller can be sent, appearing like normal voicemail but without recording any message.

16.2 Alerting System Users to New Voice Mail Messages

The Call Management System alerts each system user to the presence ofnew voice mail messages. The alerts are subscreens of the “Voice Mail”part of the “Message Board” of the call management window 115, asdescribed in Section 8. Included are:

1. The name of the caller

2. The date and time of the call

3. The length of voice mail message left

4. The telephone number of the caller.

The user may go to the voice mail system to hear his messages and/or hemay return the call directly from the voice mail subscreen by a click ofhis mouse, or he may delete the notification about the voice mailmessage.

For Call Management System integrated voice mail subsystems (see below)and other voice mail systems which can integrate with digital network(s)and computer-based systems, the Call Management System providesadditional features of:

1. Alerting system users to the presence of all new voice mail messages,not just those received through the Call Management System, includingthose calls for which no message was left.

2. Retrieving the voice mail messages directly following mouse-clickselection from the voice mail alert screen.

3. Message selection can be made by the user in any order.

4. Having the Call Management System establish a voice pathway to theuser and then instructing the voice mail system to play out the messagesselected through that voice pathway.

5. Returning calls by the click of the mouse.

16.3 Integrated Voice Mail Subsystem

When the Call Management System provides the integrated voice mail forthe organization, the voice mail is tightly integrated with the rest ofthe system, unlike other voice mail systems. Calls transferred to theintegrated voice mail are provided the usual array of voice mail callerfeatures, controlled by entry using the telephone keypad. Voice messagesare stored either as part of the call management database 215 or as partof the organization's e-mail or other message storage capabilities 110.

System users are alerted to and may review and activate their voice mailmessages from their voice mail alert screen in any order, knowing whoeach caller is and the length of their voice mail message. When a voicemail message is selected to be heard by the double-click of a mouse orotherwise, the call management computer 101 creates a voice pathway tothe user's telephone instrument 106, if one is not already present,which it then uses to play back the selected voice mail messages fromthe call management database or other storage. During playback, many newcapabilities are provided the user including:

1. Knowing the identity of the caller and the length of each voice mailmessage

2. Selecting voice mail messages to be heard in any order

3. Playback controls over speedup and slowdown, backup, fast forward,fast reverse, and many others which have limited availability withconventional voice mail systems

4. Returning the calls with the click of the mouse

5. Many others:

With integrated voice mail, the user can send a call to voice mail and,once free from other tasks, retrieve that call.

Integrated voice mail removes the limitations and barriers of existingvoice mail systems, affording system users much more information andentirely new capabilities for its use.

17. User Status

The current dynamic status of all system users is made available toother system users through their call management window. Knowing thecurrent status of users improves the ability of members of anorganization to communicate both within the organization and externallywith their customers and prospects.

The user status of all system users ill, 113 is continuously maintainedby the system through the user's call management window 115 on theirworkstation 114 and the call management database 215, as described inSection 8. The system user may change his status at any time, selectingamong a variety of status conditions:

1. “Available to receive all calls”,

2. “Available only for VIP calls”,

3. “Unavailable—transferred to another workstation”,

4. “Unavailable for all calls”,

5. and others.

When the user selects the item 4 “unavailable” option, he is given alist of potential reasons from which he may choose, e.g., “Out of theOffice”, “Out to lunch”, “On vacation”, etc. He may choose one of these,choose one and modify it or type in anything else he wishes, e.g., “Awayfrom my desk til 2:00”, “Out of the office back Friday”, “Giving a demotil 10:00” etc.

When the user is in one of the “available” states, the systemautomatically applies appropriate status, e.g., “On the phone”, “Notresponding to calls”, et

Whenever a system user needs to transfer a call, conference a call,contact another user or any number of other reasons, the call managementwindow's “Directory” as accessed in a variety of ways, (see Section 8)provides the means to determine the current status of other systemusers. Obviously, if another user is not available, there is no usetrying to transfer or conference that person with the existing call.

18. Fault Tolerance and “Copper Bypass”

Call Management System fault tolerance is accomplished in two ways,“copper bypass” and “dual system” configurations.

18.1 “Copper Bypass” Fault Tolerance

For Call Management Systems having the same kind and number of trunks onboth the CO and PBX sides 102, 202, 105, 205 “copper bypass” faulttolerance is provided. “Copper bypass” is implemented through anexternal set of physical switches 1001, FIG. 10 a, which are arranged sothat, when deactivated, the CO trunks “bypass” the Call ManagementSystem altogether, connecting directly to the PBX trunks, removing thecall management computer 101, 201 from the configuration, connecting theCO directly to the PBX.

The switches in the copper bypass box are normally energized by a signalfrom the DSP motherboard 1008, connecting the CO and PBX trunks to thecall management computer 101, 201. The normal energized state continuesso long as:

1. The power to the call management computer 101, 201 remains

2. The call management computer 101, 201 continues to operate in anappropriate manner and to refresh the switch control circuit on aregular basis, at least every few seconds

3. The DSP processors continue to operate as programmed.

If any of these conditions fail, the switches are deactivated and thesystem is instantly removed from connection to the CO and PBX trunks andthe trunks are bridged together, attaching the CO 103 to theorganization's PBX 104 as before, allowing telephone service to berestored, but without the new Call Management System features.

18.2 “Dual-System” Fault Tolerance

Dual-system fault tolerance is used for configurations in which CO trunkto PBX trunk conversions are implemented, or for configurationsrequiring a very high degree of up-time reliability where essentially nodown-time is acceptable.

Dual-system fault tolerance is provided through implementing dual callmanagement computers 101 with their trunk interface boards 203, 206,207, telephony buses 210, circuit switches 204, DSP processors 208,digital network connections 209 and databases 215. During normaloperations, one of the computers is connected to the CO and PBX trunks202, 205 and is providing the Call Management System features. The otherbackup system is also alive, attached to the data pathway 109 but not tothe CO and PBX trunks. Both systems remain alive during normaloperations in order to maintain equality of the two copies of thedatabases via the digital network. An alternative to this process is tokeep the Call Management Databases 215 on the LAN server 110 orelsewhere instead of on the call management computer 101.

When the primary system fails, the backup system is switched in-line inits place allowing business operations to continue while the othersystem is repaired and placed back in service. This process requires theCall Management Databases 215 to be “equalized” prior to the repairedsystem being placed on “backup” status, assuming they are kept on thecall management computer 101, 201. Switching the CO and PBX trunks fromthe “primary” system to the “Backup” system is done automatically usingvariations of the copper bypass boxes, as shown in FIG. 10 b.

1. A call management system for an organization comprising: at least oneuser position, comprising a computer workstation and a telephoneapparatus that is associated with the computer workstation; a callmanagement computer comprising a memory; and a digital data network toconnect the computer workstation with the call management computer, thememory, to store a plurality of call processing rules that determine howa call, directed to a user, is to be processed, the plurality of callprocessing rules being defined by the computer workstation before thecall is received, the call management computer to intercept the calldirectly from a public telephone provider's central office and beforeany telephone switch of the organization, that is incoming, to a firstuser position that is included in the at least one user position, thecall management computer to determine that the call is for the firstuser position, the call management computer to monitor the call todetermine call content and to interact with the memory and theworkstation of the first user position during the call to determine howthe call is processed based on the plurality of call processing rulesand the call content, the call management computer to process the callaccording to instructions of at least one applicable call processingrule from the workstation that is included in the plurality of callprocessing rules thereby providing call control through the computerworkstation.
 2. The call management system of claim 1, wherein thedigital data network includes at least one of a local area network, awide area network, the Internet, and an Integrated Services Digital(ISDN) network.
 3. The call management system of claim 1, wherein thecall management computer includes an apparatus to switch the call to adestination selected by the user and to tag the call with a digitalmessage which remains with the call during transfer to another user. 4.The call management system of claim 3, wherein the apparatus to switchthe call includes at least one external telephone switch.
 5. The callmanagement system of claim 4, wherein the at least one externaltelephone switch includes any one from a group including a privatebranch exchange, a Private Branch Exchange (PBX) switch, a telephone keysystem, an automatic call distributor, an Automatic Call Distribution(ACD) switch, and a telephone central office.
 6. The call managementsystem of claim 3, wherein the apparatus to switch the call includes aswitching apparatus contained within the call management computer. 7.The call management system of claim 1, wherein the at least oneapplicable call processing rule is applicable to at least one user, thatincludes the user, and determines, at least in part, how the call to theat least one user is processed.
 8. The call management system of claim7, wherein the call management system includes storage for the at leastone applicable call processing rule.
 9. The call management system ofclaim 7, wherein the at least one applicable call processing rule isdetermined to be applicable, at least in part, by an identity of theuser.
 10. The call management system of claim 7, wherein the at leastone applicable call processing rule is determined to be applicable, atleast in part, by a current status of the user.
 11. The call managementsystem of claim 10, wherein the current status of the user includes atleast one of a group including whether the user is on a phone, whetherthe user is available to receive the call, whether the user is to acceptonly a priority call, and a current location of the user.
 12. The callmanagement system of claim 7, wherein the at least one applicable callprocessing rule is determined to be applicable, at least in part, by adate, a day of the week, and a time of day.
 13. The call managementsystem of claim 7, wherein the at least one applicable call processingrule includes instructions for routing calls from at least one caller toa destination other than the first user position.
 14. The callmanagement system of claim 13, wherein the destination other than thefirst user position is at least one of a group including on the publicswitched telephone network, a second user position that is included inthe at least one user position, a destination on the Internet, and avoice mailbox.
 15. The call management system of claim 7, wherein theplurality of call processing rules includes a call processing rule thatincludes an instruction to play a prerecorded message to a caller. 16.The call management system of claim 13, wherein an identity of the atleast one user determines, at least in part, the pre-recorded messagethat is played.
 17. The call management system of claim 15, wherein thepre-recorded message requests the caller to enter information.
 18. Thecall management system of claim 17, wherein the entered information is aDual-tone Multifrequency (DTMF) signal.
 19. The call management systemof claim 17, wherein the entered information is a spoken word.
 20. Thecall management system of claim 17, wherein the entered information isused to determine, at least in part, a subsequent process for the call.21. The call management system of claim 17, wherein a second callprocessing rule, that is included in the plurality of call processingrules, specifies, at least in part, that the call is transferred to another than normal user position.
 22. The call management system of claim21, further including a transfer user function to change a location towhich the call is to be transferred.
 23. The call management system ofclaim 21, wherein the second call processing rule specifies at least onefrom a group including a series of alternate destinations which are tobe called, at least in part that the call be transferred to a pagingservice, at least in part that the call be transferred to a voicemailbox, at least in part that the call be transferred to an alternatelocation and subsequently transferred back to the at least one user, atleast in part that another call processing rule should be applied to thecall, at least in part that a special ringing sound should be used forthe call, at least in part that the call should be disconnected, and atleast in part that the call should be placed on hold.
 24. The callmanagement system of claim 13, wherein the destination other than thefirst user position includes any one from a group including a seconduser position that is included in the at least one user position and anextension number.
 25. The call management system of claim 7, wherein tointeract with the memory includes to interact with the computerworkstation that is associated with the first user position, wherein tointeract with the computer workstation includes a command to apply acall processing rule, that is included in the plurality of callprocessing rules, to the call.
 26. The call management system of claim1, further including at least one transfer button displayed on thecomputer workstation to command the call to be transferred to adestination associated with the at least one transfer button.
 27. Thecall management system of claim 1, wherein the telephone apparatus isintegrated with the computer workstation and the call managementcomputer creates a reusable voice pathway to the first user which isused for the call and all subsequent calls to the first user until allsuch calls have been processed.
 28. The call management system of claim1, wherein a call received for a non-system user is routed to anappropriate destination without further processing.
 29. The callmanagement system of claim 1, wherein the call management computerreceives a remote call from a remote destination, determines thetelephone number of the remote destination, and originates an outgoingcall to the remote destination.
 30. A call management method comprising:storing in a memory a plurality of call processing rules that determinehow a call, directed to a user, is processed, the plurality of callprocessing rules being defined by a computer workstation before the callis received; intercepting the call directly from a central office beforeany local telephone switch, that is incoming, to a first user positionthat is included in at least one user position comprising the computerworkstation and a telephone apparatus that is associated with thecomputer workstation; determining the call is for the first userposition; determining how the call is processed based on the pluralityof call processing rules and at least one instruction from the computerworkstation during the call; and processing the call according toinstructions of at least one applicable call processing rule from theworkstation that is included in the plurality of call processing rules.31. The call management method of claim 30, wherein the at least oneapplicable call processing rule is applicable to at least one user, thatincludes the user, and determines, at least in part, how the call to theat least one user is processed.
 32. The call management method of claim31, further including determining the at least one applicable callprocessing rule is applicable based on any one from a group including anidentity of the user and a current status of the user.
 33. The callmanagement method of claim 32, wherein the current status of the userincludes any one from a group including whether the user is on a phone,whether the user is available to receive the call, whether the useraccepts only a priority call, and a location of the user.
 34. The callmanagement method of claim 31, further including determining the atleast one applicable call processing rule is applicable based on acurrent date, a day of a week and a time of day.
 35. The call managementmethod of claim 31, wherein the at least one applicable call processingrule includes instructions for routing the call from at least one callerto a destination other than the first user position.
 36. The callmanagement method of claim 35, wherein the destination other than thefirst user position is any one from a group including a destination onthe public switched telephone network, a second user position that isincluded in the at least one user position, a destination on theInternet, and a voice mailbox.
 37. The call management method of claim31, wherein the plurality of call processing rules includes a callprocessing rule that includes an instruction to play a pre-recordedmessage to a caller.
 38. The call management method of claim 36, furtherincluding determining the pre-recorded message that is played based onan identity of the at least one user.
 39. The call management method ofclaim 37, wherein the pre-recorded message requests the caller to enterinformation.
 40. The call management method of claim 39, wherein theentered information is any one from a group including a Dual-toneMultifrequency (DTMF) signal and a spoken word.
 41. The call managementmethod of claim 39, wherein the entered information determines at least,in part, a subsequent process for the call.
 42. The call managementmethod of claim 31, wherein a second call processing rule, that isincluded in the plurality of call processing rules, specifies, at leastin part, that the call is transferred to an other than normal userposition.
 43. The call management method of claim 30, further includingchanging a location to which the call is to be transferred.
 44. The callmanagement method of claim 30, wherein the plurality of call processingrules includes a call processing rule that specifies a series ofalternate destinations which are to be called.
 45. The call managementmethod of claim 42, wherein the second call processing rule specifiesany one from a group including at least in part that the call betransferred to a paging service, at least in part that the call betransferred to a voice mailbox, at least in part that the call betransferred to an alternate location and subsequently transferred backto the at least one user, at least in part that another call processingrule should be applied to the call, at least in part that a specialringing sound should be used for the call, at least in part that thecall should be disconnected, at least in part that the call should beplaced on hold.
 46. The call management method of claim 35, wherein thedestination other than the first user position is any one from a groupincluding a second user position that is included in the at least oneuser position and identified with an extension number.
 47. The callmanagement method of claim 31, wherein determining how the call is to beprocessed includes interacting with the computer workstation that isassociated with the first user position, wherein interacting with thecomputer workstation includes a command to apply a call processing rulethat is included in the plurality of call processing rules to the call.48. The call management method of claim 31, further includingtransferring the call to a destination associated with at least onetransfer button.
 49. The call management method of claim 31, wherein thetelephone apparatus is integrated with the computer workstation.
 50. Thecall management method of claim 31, further including receiving a callfor a non-system user and routing the call to an appropriate destinationwithout further processing.
 51. The call management method of claim 31,further including receiving a remote call from a remote destination,determining a telephone number of the remote destination, andoriginating an outgoing call to the remote destination.
 52. Anon-transitory tangible machine readable medium storing a set ofinstructions that, when executed by a machine, cause the machine to:store a plurality of call processing rules that determine how a call,directed to a user, is to be processed, the plurality of call processingrules being defined by a computer workstation before the call isreceived; intercept the call directly from the central office and beforeany local telephone switch, that is incoming to a first user positionthat is included in at least one user position comprising the computerworkstation and a telephone apparatus that is associated with thecomputer workstation; determine the call is for the first user position;determine how the call is to be processed based on the plurality of callprocessing rules and at least one instruction from the computerworkstation during the call; and process the call according toinstructions of at least one applicable call processing rule from theworkstation that is included in the plurality of call processing rules.